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May 17th, 2010
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  1. Asterisk 1.4.26.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
  2. Created by Mark Spencer <[email protected]>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 1.4.26.1 currently running on meetmecall-01 (pid = 2685)
  9. meetmecall-01*CLI>
  10. Core debug is at least 10
  11.  
  12. [Kmeetmecall-01*CLI>
  13. [May 17 20:04:06] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for [email protected]
  14. REGISTER 13 headers, 0 lines
  15. REGISTER attempt 1 to [email protected]
  16. Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
  17. REGISTER sip:voip1.sig.itsp.nl SIP/2.0
  18.  
  19. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;rport
  20.  
  21. From: <sip:[email protected]>;tag=as2bcd00dc
  22.  
  23. To: <sip:[email protected]>
  24.  
  25. Call-ID: [email protected]
  26.  
  27. CSeq: 160 REGISTER
  28.  
  29. User-Agent: Asterisk PBX
  30.  
  31. Max-Forwards: 70
  32.  
  33. Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:voip1.sig.itsp.nl", nonce="3929cf94", response="bb5acb40a340de5f14bcc63f985d3051"
  34.  
  35. Expires: 120
  36.  
  37. Contact: <sip:[email protected]>
  38.  
  39. Event: registration
  40.  
  41. Content-Length: 0
  42.  
  43.  
  44.  
  45.  
  46. ---
  47.  
  48. [Kmeetmecall-01*CLI>
  49.  
  50. <--- SIP read from xx.xxx.204.68:5060 --->
  51. SIP/2.0 100 Trying
  52.  
  53. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;received=xx.xx.241.180;rport=5060
  54.  
  55. From: <sip:[email protected]>;tag=as2bcd00dc
  56.  
  57. To: <sip:[email protected]>
  58.  
  59. Call-ID: [email protected]
  60.  
  61. CSeq: 160 REGISTER
  62.  
  63. User-Agent: itsp VoIP Interconnect
  64.  
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  66.  
  67. Supported: replaces
  68.  
  69. Content-Length: 0
  70.  
  71.  
  72.  
  73.  
  74. <------------->
  75. --- (10 headers 0 lines) ---
  76.  
  77. [Kmeetmecall-01*CLI>
  78.  
  79. <--- SIP read from xx.xxx.204.68:5060 --->
  80. SIP/2.0 401 Unauthorized
  81.  
  82. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;received=xx.xx.241.180;rport=5060
  83.  
  84. From: <sip:[email protected]>;tag=as2bcd00dc
  85.  
  86. To: <sip:[email protected]>;tag=as343649a4
  87.  
  88. Call-ID: [email protected]
  89.  
  90. CSeq: 160 REGISTER
  91.  
  92. User-Agent: itsp VoIP Interconnect
  93.  
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  95.  
  96. Supported: replaces
  97.  
  98. WWW-Authenticate: Digest algorithm=MD5, realm="itsp.nl", nonce="4b514668"
  99.  
  100. Content-Length: 0
  101.  
  102.  
  103.  
  104.  
  105. <------------->
  106. --- (11 headers 0 lines) ---
  107.  
  108. [Kmeetmecall-01*CLI>
  109. Responding to challenge, registration to domain/host name voip1.sig.itsp.nl
  110.  
  111. [Kmeetmecall-01*CLI>
  112. REGISTER 13 headers, 0 lines
  113. REGISTER attempt 2 to [email protected]
  114.  
  115. [Kmeetmecall-01*CLI>
  116. Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
  117. REGISTER sip:voip1.sig.itsp.nl SIP/2.0
  118.  
  119. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;rport
  120.  
  121. From: <sip:[email protected]>;tag=as280729a5
  122.  
  123. To: <sip:[email protected]>
  124.  
  125. Call-ID: [email protected]
  126.  
  127. CSeq: 161 REGISTER
  128.  
  129. User-Agent: Asterisk PBX
  130.  
  131. Max-Forwards: 70
  132.  
  133. Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:voip1.sig.itsp.nl", nonce="4b514668", response="5bbce326e107aed7d02d4b3ed205d421"
  134.  
  135. Expires: 120
  136.  
  137. Contact: <sip:[email protected]>
  138.  
  139. Event: registration
  140.  
  141. Content-Length: 0
  142.  
  143.  
  144.  
  145.  
  146. ---
  147.  
  148. [Kmeetmecall-01*CLI>
  149.  
  150. <--- SIP read from xx.xxx.204.68:5060 --->
  151. SIP/2.0 100 Trying
  152.  
  153. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;received=xx.xx.241.180;rport=5060
  154.  
  155. From: <sip:[email protected]>;tag=as280729a5
  156.  
  157. To: <sip:[email protected]>
  158.  
  159. Call-ID: [email protected]
  160.  
  161. CSeq: 161 REGISTER
  162.  
  163. User-Agent: itsp VoIP Interconnect
  164.  
  165. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  166.  
  167. Supported: replaces
  168.  
  169. Content-Length: 0
  170.  
  171.  
  172.  
  173.  
  174. <------------->
  175. --- (10 headers 0 lines) ---
  176.  
  177. [Kmeetmecall-01*CLI>
  178.  
  179. <--- SIP read from xx.xxx.204.68:5060 --->
  180. SIP/2.0 200 OK
  181.  
  182. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;received=xx.xx.241.180;rport=5060
  183.  
  184. From: <sip:[email protected]>;tag=as280729a5
  185.  
  186. To: <sip:[email protected]>;tag=as343649a4
  187.  
  188. Call-ID: [email protected]
  189.  
  190. CSeq: 161 REGISTER
  191.  
  192. User-Agent: itsp VoIP Interconnect
  193.  
  194. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  195.  
  196. Supported: replaces
  197.  
  198. Expires: 120
  199.  
  200. Contact: <sip:[email protected]>;expires=120
  201.  
  202. Date: Mon, 17 May 2010 18:02:48 GMT
  203.  
  204. Content-Length: 0
  205.  
  206.  
  207.  
  208.  
  209. <------------->
  210. --- (13 headers 0 lines) ---
  211.  
  212. [Kmeetmecall-01*CLI>
  213. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  214.  
  215. [Kmeetmecall-01*CLI>
  216. [May 17 20:04:06] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for voip1.sig.itsp.nl is 120 sec (Scheduling reregistration in 105 s)
  217.  
  218. [Kmeetmecall-01*CLI>
  219. Really destroying SIP dialog '[email protected]' Method: REGISTER
  220.  
  221. [Kmeetmecall-01*CLI>
  222. Really destroying SIP dialog '[email protected]' Method: REGISTER
  223.  
  224. [Kmeetmecall-01*CLI>
  225. Reliably Transmitting (NAT) to xx.xx.71.170:1030:
  226. OPTIONS sip:[email protected]:5060 SIP/2.0
  227.  
  228. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK2653b67f;rport
  229.  
  230. From: "asterisk" <sip:[email protected]>;tag=as07a56e9e
  231.  
  232. To: <sip:[email protected]:5060>
  233.  
  234. Contact: <sip:[email protected]>
  235.  
  236. Call-ID: [email protected]
  237.  
  238. CSeq: 102 OPTIONS
  239.  
  240. User-Agent: Asterisk PBX
  241.  
  242. Max-Forwards: 70
  243.  
  244. Date: Mon, 17 May 2010 18:04:11 GMT
  245.  
  246. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  247.  
  248. Supported: replaces
  249.  
  250. Content-Length: 0
  251.  
  252.  
  253.  
  254.  
  255. ---
  256.  
  257. [Kmeetmecall-01*CLI>
  258.  
  259. <--- SIP read from xx.xx.71.170:1030 --->
  260. SIP/2.0 200 OK
  261.  
  262. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK2653b67f;rport
  263.  
  264. From: "asterisk" <sip:[email protected]>;tag=as07a56e9e
  265.  
  266. To: <sip:[email protected]:5060>;tag=602B828B-603E4C74
  267.  
  268. CSeq: 102 OPTIONS
  269.  
  270. Call-ID: [email protected]
  271.  
  272. Contact: <sip:[email protected]:5060>
  273.  
  274. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  275.  
  276. User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
  277.  
  278.  
  279. [Kmeetmecall-01*CLI>
  280. Content-Length: 0
  281.  
  282.  
  283.  
  284.  
  285. <------------->
  286. --- (10 headers 0 lines) ---
  287.  
  288. [Kmeetmecall-01*CLI>
  289. Really destroying SIP dialog '[email protected]' Method: OPTIONS
  290.  
  291. [Kmeetmecall-01*CLI>
  292. Reliably Transmitting (NAT) to xx.xx.71.170:5060:
  293. OPTIONS sip:[email protected]:5060 SIP/2.0
  294.  
  295. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK04978a6e;rport
  296.  
  297. From: "asterisk" <sip:[email protected]>;tag=as773a250d
  298.  
  299. To: <sip:[email protected]:5060>
  300.  
  301. Contact: <sip:[email protected]>
  302.  
  303. Call-ID: [email protected]
  304.  
  305. CSeq: 102 OPTIONS
  306.  
  307. User-Agent: Asterisk PBX
  308.  
  309. Max-Forwards: 70
  310.  
  311. Date: Mon, 17 May 2010 18:04:13 GMT
  312.  
  313. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  314.  
  315. Supported: replaces
  316.  
  317. Content-Length: 0
  318.  
  319.  
  320.  
  321.  
  322. ---
  323.  
  324. [Kmeetmecall-01*CLI>
  325.  
  326. <--- SIP read from xx.xx.71.170:5060 --->
  327. SIP/2.0 200 OK
  328.  
  329. From: "asterisk"<sip:[email protected]>;tag=as773a250d
  330.  
  331. To: <sip:[email protected]:5060>;tag=6f01a8c0-13c4-3c9d2-ecbd2dc-bcf
  332.  
  333. Call-ID: [email protected]
  334.  
  335. CSeq: 102 OPTIONS
  336.  
  337. Via: SIP/2.0/UDP xx.xx.241.180:5060;rport=5060;branch=z9hG4bK04978a6e
  338.  
  339. Supported: replaces,100rel,timer
  340.  
  341. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS
  342.  
  343. User-Agent: Swissvoice IP10S
  344.  
  345. Accept: application/sdp
  346.  
  347. Content-Length: 0
  348.  
  349.  
  350.  
  351.  
  352. <------------->
  353. --- (11 headers 0 lines) ---
  354. Really destroying SIP dialog '[email protected]' Method: OPTIONS
  355.  
  356. [Kmeetmecall-01*CLI>
  357. [May 17 20:04:16] NOTICE[3361]: pbx_spool.c:366 attempt_thread: Call completed to Local/s@gtalk_recording_local_1/n
  358.  
  359. [Kmeetmecall-01*CLI>
  360. Audio is at xx.xx.241.180 port 17186
  361. Adding codec 0x8 (alaw) to SDP
  362.  
  363. [Kmeetmecall-01*CLI>
  364. Adding non-codec 0x1 (telephone-event) to SDP
  365.  
  366. [Kmeetmecall-01*CLI>
  367. Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
  368. INVITE sip:[email protected] SIP/2.0
  369.  
  370. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;rport
  371.  
  372. From: "asterisk" <sip:[email protected]>;tag=as74578279
  373.  
  374. To: <sip:[email protected]>
  375.  
  376. Contact: <sip:[email protected]>
  377.  
  378. Call-ID: [email protected]
  379.  
  380. CSeq: 102 INVITE
  381.  
  382. User-Agent: Asterisk PBX
  383.  
  384. Max-Forwards: 70
  385.  
  386. Date: Mon, 17 May 2010 18:04:17 GMT
  387.  
  388. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  389.  
  390. Supported: replaces
  391.  
  392. Content-Type: application/sdp
  393.  
  394. Content-Length: 240
  395.  
  396.  
  397.  
  398. v=0
  399.  
  400. o=root 2685 2685 IN IP4 xx.xx.241.180
  401.  
  402. s=session
  403.  
  404. c=IN IP4 xx.xx.241.180
  405.  
  406. t=0 0
  407.  
  408. m=audio 17186 RTP/AVP 8 101
  409.  
  410. a=rtpmap:8 PCMA/8000
  411.  
  412. a=rtpmap:101 telephone-event/8000
  413.  
  414. a=fmtp:101 0-16
  415.  
  416. a=silenceSupp:off - - - -
  417.  
  418. a=ptime:20
  419.  
  420. a=sendrecv
  421.  
  422.  
  423. ---
  424.  
  425. [Kmeetmecall-01*CLI>
  426.  
  427. <--- SIP read from xx.xxx.204.68:5060 --->
  428. SIP/2.0 407 Proxy Authentication Required
  429.  
  430. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;received=xx.xx.241.180;rport=5060
  431.  
  432. From: "asterisk" <sip:[email protected]>;tag=as74578279
  433.  
  434. To: <sip:[email protected]>;tag=as6a8dee5b
  435.  
  436. Call-ID: [email protected]
  437.  
  438. CSeq: 102 INVITE
  439.  
  440. User-Agent: itsp VoIP Interconnect
  441.  
  442. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  443.  
  444. Supported: replaces
  445.  
  446. Proxy-Authenticate: Digest algorithm=MD5, realm="itsp.nl", nonce="3d0e6e4f"
  447.  
  448. Content-Length: 0
  449.  
  450.  
  451.  
  452.  
  453. <------------->
  454. --- (11 headers 0 lines) ---
  455.  
  456. [Kmeetmecall-01*CLI>
  457. Transmitting (no NAT) to xx.xxx.204.68:5060:
  458. ACK sip:[email protected] SIP/2.0
  459.  
  460. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;rport
  461.  
  462. From: "asterisk" <sip:[email protected]>;tag=as74578279
  463.  
  464. To: <sip:[email protected]>;tag=as6a8dee5b
  465.  
  466. Contact: <sip:[email protected]>
  467.  
  468. Call-ID: [email protected]
  469.  
  470. CSeq: 102 ACK
  471.  
  472. User-Agent: Asterisk PBX
  473.  
  474. Max-Forwards: 70
  475.  
  476. Content-Length: 0
  477.  
  478.  
  479.  
  480.  
  481. ---
  482.  
  483. [Kmeetmecall-01*CLI>
  484. Audio is at xx.xx.241.180 port 17186
  485. Adding codec 0x8 (alaw) to SDP
  486.  
  487. [Kmeetmecall-01*CLI>
  488. Adding non-codec 0x1 (telephone-event) to SDP
  489.  
  490. [Kmeetmecall-01*CLI>
  491. Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
  492. INVITE sip:[email protected] SIP/2.0
  493.  
  494. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;rport
  495.  
  496. From: "asterisk" <sip:[email protected]>;tag=as74578279
  497.  
  498. To: <sip:[email protected]>
  499.  
  500. Contact: <sip:[email protected]>
  501.  
  502. Call-ID: [email protected]
  503.  
  504. CSeq: 103 INVITE
  505.  
  506. User-Agent: Asterisk PBX
  507.  
  508. Max-Forwards: 70
  509.  
  510. Proxy-Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:[email protected]", nonce="3d0e6e4f", response="b244bdd1e2d5cde5a8b309f3f5bf48fc"
  511.  
  512. Date: Mon, 17 May 2010 18:04:17 GMT
  513.  
  514. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  515.  
  516. Supported: replaces
  517.  
  518. Content-Type: application/sdp
  519.  
  520. Content-Length: 240
  521.  
  522.  
  523.  
  524. v=0
  525.  
  526. o=root 2685 2686 IN IP4 xx.xx.241.180
  527.  
  528. s=session
  529.  
  530. c=IN IP4 xx.xx.241.180
  531.  
  532. t=0 0
  533.  
  534. m=audio 17186 RTP/AVP 8 101
  535.  
  536. a=rtpmap:8 PCMA/8000
  537.  
  538. a=rtpmap:101 telephone-event/8000
  539.  
  540. a=fmtp:101 0-16
  541.  
  542. a=silenceSupp:off - - - -
  543.  
  544. a=ptime:20
  545.  
  546. a=sendrecv
  547.  
  548.  
  549. ---
  550.  
  551. [Kmeetmecall-01*CLI>
  552.  
  553. <--- SIP read from xx.xxx.204.68:5060 --->
  554. SIP/2.0 100 Trying
  555.  
  556. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
  557.  
  558. From: "asterisk" <sip:[email protected]>;tag=as74578279
  559.  
  560. To: <sip:[email protected]>
  561.  
  562. Call-ID: [email protected]
  563.  
  564. CSeq: 103 INVITE
  565.  
  566. User-Agent: itsp VoIP Interconnect
  567.  
  568. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  569.  
  570. Supported: replaces
  571.  
  572. Contact: <sip:[email protected]>
  573.  
  574. Content-Length: 0
  575.  
  576.  
  577.  
  578.  
  579. <------------->
  580. --- (11 headers 0 lines) ---
  581.  
  582. [Kmeetmecall-01*CLI>
  583. Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
  584. OPTIONS sip:xx.xxx.204.68 SIP/2.0
  585.  
  586. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK47198280;rport
  587.  
  588. From: "asterisk" <sip:[email protected]>;tag=as12b9ca60
  589.  
  590. To: <sip:xx.xxx.204.68>
  591.  
  592. Contact: <sip:[email protected]>
  593.  
  594. Call-ID: [email protected]
  595.  
  596. CSeq: 102 OPTIONS
  597.  
  598. User-Agent: Asterisk PBX
  599.  
  600. Max-Forwards: 70
  601.  
  602. Date: Mon, 17 May 2010 18:04:21 GMT
  603.  
  604. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  605.  
  606. Supported: replaces
  607.  
  608. Content-Length: 0
  609.  
  610.  
  611.  
  612.  
  613. ---
  614.  
  615. [Kmeetmecall-01*CLI>
  616.  
  617. <--- SIP read from xx.xxx.204.68:5060 --->
  618. SIP/2.0 404 Not Found
  619.  
  620. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK47198280;received=xx.xx.241.180;rport=5060
  621.  
  622. From: "asterisk" <sip:[email protected]>;tag=as12b9ca60
  623.  
  624. To: <sip:xx.xxx.204.68>;tag=as64a1479f
  625.  
  626. Call-ID: [email protected]
  627.  
  628. CSeq: 102 OPTIONS
  629.  
  630. User-Agent: itsp VoIP Interconnect
  631.  
  632. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  633.  
  634. Supported: replaces
  635.  
  636. Accept: application/sdp
  637.  
  638. Content-Length: 0
  639.  
  640.  
  641.  
  642. [Kmeetmecall-01*CLI>
  643.  
  644.  
  645. <------------->
  646. --- (11 headers 0 lines) ---
  647.  
  648. [Kmeetmecall-01*CLI>
  649. Really destroying SIP dialog '[email protected]' Method: OPTIONS
  650.  
  651. [Kmeetmecall-01*CLI>
  652. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  653. OPTIONS sip:itsp2.net:6060 SIP/2.0
  654.  
  655. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK4a1aadc7;rport
  656.  
  657. From: "asterisk" <sip:[email protected]>;tag=as0dd6cca1
  658.  
  659. To: <sip:itsp2.net:6060>
  660.  
  661. Contact: <sip:[email protected]>
  662.  
  663. Call-ID: [email protected]
  664.  
  665. CSeq: 102 OPTIONS
  666.  
  667. User-Agent: Asterisk PBX
  668.  
  669. Max-Forwards: 70
  670.  
  671. Date: Mon, 17 May 2010 18:04:21 GMT
  672.  
  673. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  674.  
  675. Supported: replaces
  676.  
  677. Content-Length: 0
  678.  
  679.  
  680.  
  681.  
  682. ---
  683.  
  684. [Kmeetmecall-01*CLI>
  685.  
  686. <--- SIP read from xx.xxx.119.38:6060 --->
  687. SIP/2.0 200 OK
  688.  
  689. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK4a1aadc7;rport
  690.  
  691. From: "asterisk" <sip:[email protected]>;tag=as0dd6cca1
  692.  
  693. To: <sip:itsp2.net:6060>
  694.  
  695. Call-ID: [email protected]
  696.  
  697. CSeq: 102 OPTIONS
  698.  
  699. Contact: <sip:xx.xxx.119.38:6060>
  700.  
  701. User-Agent: Asterisk PBX
  702.  
  703. Date: Mon, 17 May 2010 18:04:21 GMT
  704.  
  705. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  706.  
  707. Accept: application/sdp, application/qsig, text/plain, application/dtmf-relay
  708.  
  709. Content-Length: 0
  710.  
  711.  
  712.  
  713.  
  714. <------------->
  715. --- (12 headers 0 lines) ---
  716. Really destroying SIP dialog '[email protected]' Method: OPTIONS
  717.  
  718. [Kmeetmecall-01*CLI>
  719. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  720. OPTIONS sip:itsp2.net:6060 SIP/2.0
  721.  
  722. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK11453973;rport
  723.  
  724. From: "asterisk" <sip:[email protected]>;tag=as0d16bc6c
  725.  
  726. To: <sip:itsp2.net:6060>
  727.  
  728. Contact: <sip:[email protected]>
  729.  
  730. Call-ID: [email protected]
  731.  
  732. CSeq: 102 OPTIONS
  733.  
  734. User-Agent: Asterisk PBX
  735.  
  736. Max-Forwards: 70
  737.  
  738. Date: Mon, 17 May 2010 18:04:21 GMT
  739.  
  740. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  741.  
  742. Supported: replaces
  743.  
  744. Content-Length: 0
  745.  
  746.  
  747.  
  748.  
  749. ---
  750.  
  751. [Kmeetmecall-01*CLI>
  752.  
  753. <--- SIP read from xx.xxx.119.38:6060 --->
  754. SIP/2.0 200 OK
  755.  
  756. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK11453973;rport
  757.  
  758. From: "asterisk" <sip:[email protected]>;tag=as0d16bc6c
  759.  
  760. To: <sip:itsp2.net:6060>
  761.  
  762. Call-ID: [email protected]
  763.  
  764. CSeq: 102 OPTIONS
  765.  
  766. Contact: <sip:xx.xxx.119.38:6060>
  767.  
  768. User-Agent: Asterisk PBX
  769.  
  770. Date: Mon, 17 May 2010 18:04:21 GMT
  771.  
  772. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  773.  
  774. Accept: application/sdp, application/qsig, text/plain, application/dtmf-relay
  775.  
  776. Content-Length: 0
  777.  
  778.  
  779.  
  780.  
  781. <------------->
  782. --- (12 headers 0 lines) ---
  783. Really destroying SIP dialog '[email protected]' Method: OPTIONS
  784.  
  785. [Kmeetmecall-01*CLI>
  786. [May 17 20:04:22] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for [email protected]
  787. REGISTER 12 headers, 0 lines
  788. REGISTER attempt 1 to [email protected]
  789. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  790. REGISTER sip:itsp2.net:6060 SIP/2.0
  791.  
  792. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK09bb30e0;rport
  793.  
  794. From: <sip:[email protected]>;tag=as646cd458
  795.  
  796. To: <sip:[email protected]>
  797.  
  798. Call-ID: [email protected]
  799.  
  800. CSeq: 170 REGISTER
  801.  
  802.  
  803. [Kmeetmecall-01*CLI>
  804. User-Agent: Asterisk PBX
  805.  
  806. Max-Forwards: 70
  807.  
  808. Expires: 120
  809.  
  810. Contact: <sip:[email protected]>
  811.  
  812. Event: registration
  813.  
  814. Content-Length: 0
  815.  
  816.  
  817.  
  818.  
  819. ---
  820.  
  821. [Kmeetmecall-01*CLI>
  822.  
  823. <--- SIP read from xx.xxx.119.38:6060 --->
  824. SIP/2.0 200 OK
  825.  
  826. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK09bb30e0;rport
  827.  
  828. From: <sip:[email protected]>;tag=as646cd458
  829.  
  830. To: <sip:[email protected]>
  831.  
  832. Call-ID: [email protected]
  833.  
  834. CSeq: 170 REGISTER
  835.  
  836. Contact: <sip:[email protected]>
  837.  
  838. User-Agent: Asterisk PBX
  839.  
  840. Expires: 60
  841.  
  842. Event: registration
  843.  
  844. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  845.  
  846. Content-Length: 0
  847.  
  848.  
  849.  
  850.  
  851. <------------->
  852. --- (12 headers 0 lines) ---
  853. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  854. [May 17 20:04:22] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
  855.  
  856. [Kmeetmecall-01*CLI>
  857. [May 17 20:04:22] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for [email protected]
  858. REGISTER 12 headers, 0 lines
  859. REGISTER attempt 1 to [email protected]
  860. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  861. REGISTER sip:itsp2.net:6060 SIP/2.0
  862.  
  863. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK498293f1;rport
  864.  
  865. From: <sip:[email protected]>;tag=as48d0e10b
  866.  
  867. To: <sip:[email protected]>
  868.  
  869. Call-ID: [email protected]
  870.  
  871. CSeq: 170 REGISTER
  872.  
  873.  
  874. [Kmeetmecall-01*CLI>
  875. User-Agent: Asterisk PBX
  876.  
  877. Max-Forwards: 70
  878.  
  879. Expires: 120
  880.  
  881. Contact: <sip:[email protected]>
  882.  
  883. Event: registration
  884.  
  885. Content-Length: 0
  886.  
  887.  
  888.  
  889.  
  890. ---
  891.  
  892. [Kmeetmecall-01*CLI>
  893.  
  894. <--- SIP read from xx.xxx.119.38:6060 --->
  895. SIP/2.0 200 OK
  896.  
  897. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK498293f1;rport
  898.  
  899. From: <sip:[email protected]>;tag=as48d0e10b
  900.  
  901. To: <sip:[email protected]>
  902.  
  903. Call-ID: [email protected]
  904.  
  905. CSeq: 170 REGISTER
  906.  
  907. Contact: <sip:[email protected]>
  908.  
  909. User-Agent: Asterisk PBX
  910.  
  911. Expires: 60
  912.  
  913. Event: registration
  914.  
  915. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  916.  
  917. Content-Length: 0
  918.  
  919.  
  920.  
  921.  
  922. <------------->
  923. --- (12 headers 0 lines) ---
  924. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  925. [May 17 20:04:22] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
  926.  
  927. [Kmeetmecall-01*CLI>
  928.  
  929. <--- SIP read from xx.xxx.204.68:5060 --->
  930. SIP/2.0 183 Session Progress
  931.  
  932. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
  933.  
  934. From: "asterisk" <sip:[email protected]>;tag=as74578279
  935.  
  936. To: <sip:[email protected]>;tag=as25ad1b42
  937.  
  938. Call-ID: [email protected]
  939.  
  940. CSeq: 103 INVITE
  941.  
  942. User-Agent: itsp VoIP Interconnect
  943.  
  944. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  945.  
  946. Supported: replaces
  947.  
  948. Contact: <sip:[email protected]>
  949.  
  950. Content-Type: application/sdp
  951.  
  952. Content-Length: 240
  953.  
  954.  
  955.  
  956. v=0
  957.  
  958. o=root 5032 5032 IN IP4 xx.xxx.204.68
  959.  
  960. s=session
  961.  
  962. c=IN IP4 xx.xxx.204.68
  963.  
  964. t=0 0
  965.  
  966. m=audio 14270 RTP/AVP 8 101
  967.  
  968. a=rtpmap:8 PCMA/8000
  969.  
  970. a=rtpmap:101 telephone-event/8000
  971.  
  972. a=fmtp:101 0-16
  973.  
  974. a=silenceSupp:off - - - -
  975.  
  976. a=ptime:20
  977.  
  978. a=sendrecv
  979.  
  980.  
  981. <------------->
  982. --- (12 headers 12 lines) ---
  983. Found RTP audio format 8
  984. Found RTP audio format 101
  985. Peer audio RTP is at port xx.xxx.204.68:14270
  986. Found audio description format PCMA for ID 8
  987. Found audio description format telephone-event for ID 101
  988. Got unsupported a:fmtp in SDP offer
  989. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  990. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  991. Peer audio RTP is at port xx.xxx.204.68:14270
  992.  
  993. [Kmeetmecall-01*CLI>
  994.  
  995. <--- SIP read from xx.xxx.204.68:5060 --->
  996. SIP/2.0 180 Ringing
  997.  
  998. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
  999.  
  1000. From: "asterisk" <sip:[email protected]>;tag=as74578279
  1001.  
  1002. To: <sip:[email protected]>;tag=as25ad1b42
  1003.  
  1004. Call-ID: [email protected]
  1005.  
  1006. CSeq: 103 INVITE
  1007.  
  1008. User-Agent: itsp VoIP Interconnect
  1009.  
  1010. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1011.  
  1012. Supported: replaces
  1013.  
  1014. Contact: <sip:[email protected]>
  1015.  
  1016. Content-Length: 0
  1017.  
  1018.  
  1019.  
  1020.  
  1021. <------------->
  1022. --- (11 headers 0 lines) ---
  1023.  
  1024. [Kmeetmecall-01*CLI>
  1025.  
  1026. <--- SIP read from xx.xxx.204.68:5060 --->
  1027. SIP/2.0 200 OK
  1028.  
  1029. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
  1030.  
  1031. From: "asterisk" <sip:[email protected]>;tag=as74578279
  1032.  
  1033. To: <sip:[email protected]>;tag=as25ad1b42
  1034.  
  1035. Call-ID: [email protected]
  1036.  
  1037. CSeq: 103 INVITE
  1038.  
  1039. User-Agent: itsp VoIP Interconnect
  1040.  
  1041. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1042.  
  1043. Supported: replaces
  1044.  
  1045. Contact: <sip:[email protected]>
  1046.  
  1047. C
  1048. [Kmeetmecall-01*CLI>
  1049. ontent-Type: application/sdp
  1050.  
  1051. Content-Length: 240
  1052.  
  1053.  
  1054.  
  1055. v=0
  1056.  
  1057. o=root 5032 5033 IN IP4 xx.xxx.204.68
  1058.  
  1059. s=session
  1060.  
  1061. c=IN IP4 xx.xxx.204.68
  1062.  
  1063. t=0 0
  1064.  
  1065. m=audio 14270 RTP/AVP 8 101
  1066.  
  1067. a=rtpmap:8 PCMA/8000
  1068.  
  1069. a=rtpmap:101 telephone-event/8000
  1070.  
  1071. a=fmtp:101 0-16
  1072.  
  1073. a=silenceSupp:off - - - -
  1074.  
  1075. a=ptime:20
  1076.  
  1077. a=sendrecv
  1078.  
  1079.  
  1080. <------------->
  1081. --- (12 headers 12 lines) ---
  1082. Found RTP audio format 8
  1083. Found RTP audio format 101
  1084. Peer audio RTP is at port xx.xxx.204.68:14270
  1085. Found audio description format PCMA for ID 8
  1086. Found audio description format telephone-event for ID 101
  1087. Got unsupported a:fmtp in SDP offer
  1088. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  1089. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  1090. Peer audio RTP is at port xx.xxx.204.68:14270
  1091. list_route: hop: <sip:[email protected]>
  1092. set_destination: Parsing <sip:[email protected]> for address/port to send to
  1093. set_destination: set destination to xx.xxx.204.68, port 5060
  1094. Transmitting (no NAT) to xx.xxx.204.68:5060:
  1095. ACK sip:[email protected] SIP/2.0
  1096.  
  1097. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK559d8706;rport
  1098.  
  1099. From: "asterisk" <sip:[email protected]>;tag=as74578279
  1100.  
  1101. To: <sip:[email protected]>;tag=as25ad1b42
  1102.  
  1103. Contact: <sip:[email protected]>
  1104.  
  1105. Call-ID: [email protected]
  1106.  
  1107. CSeq: 103 ACK
  1108.  
  1109. User-Agent: Asterisk PBX
  1110.  
  1111. Max-Forwards: 70
  1112.  
  1113. Content-Length: 0
  1114.  
  1115.  
  1116.  
  1117.  
  1118. ---
  1119.  
  1120. [Kmeetmecall-01*CLI>
  1121. Really destroying SIP dialog '[email protected]' Method: REGISTER
  1122.  
  1123. [Kmeetmecall-01*CLI>
  1124. Really destroying SIP dialog '[email protected]' Method: REGISTER
  1125.  
  1126. [Kmeetmecall-01*CLI>
  1127. Really destroying SIP dialog '[email protected]' Method: REGISTER
  1128.  
  1129. [Kmeetmecall-01*CLI>
  1130.  
  1131. <--- SIP read from xx.xxx.204.68:5060 --->
  1132. BYE sip:[email protected] SIP/2.0
  1133.  
  1134. Via: SIP/2.0/UDP xx.xxx.204.68:5060;branch=z9hG4bK08b7b869;rport
  1135.  
  1136. From: <sip:[email protected]>;tag=as25ad1b42
  1137.  
  1138. To: "asterisk" <sip:[email protected]>;tag=as74578279
  1139.  
  1140. Call-ID: [email protected]
  1141.  
  1142. CSeq: 102 BYE
  1143.  
  1144. User-Agent: itsp VoIP Interconnect
  1145.  
  1146. Max-Forwards: 70
  1147.  
  1148. X-Asterisk-HangupCause: Normal Clearing
  1149.  
  1150. X-Asterisk-HangupCauseCode: 16
  1151.  
  1152. Content-Length: 0
  1153.  
  1154.  
  1155.  
  1156.  
  1157. <------------->
  1158. --- (11 headers 0 lines) ---
  1159. Sending to xx.xxx.204.68 : 5060 (no NAT)
  1160.  
  1161. <--- Transmitting (no NAT) to xx.xxx.204.68:5060 --->
  1162. SIP/2.0 200 OK
  1163.  
  1164. Via: SIP/2.0/UDP xx.xxx.204.68:5060;branch=z9hG4bK08b7b869;received=xx.xxx.204.68;rport=5060
  1165.  
  1166. From: <sip:[email protected]>;tag=as25ad1b42
  1167.  
  1168. To: "asterisk" <sip:[email protected]>;tag=as74578279
  1169.  
  1170. Call-ID: [email protected]
  1171.  
  1172. CSeq: 102 BYE
  1173.  
  1174. User-Agent: Asterisk PBX
  1175.  
  1176. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1177.  
  1178. Supported: replaces
  1179.  
  1180. Content-Length: 0
  1181.  
  1182.  
  1183.  
  1184.  
  1185. <------------>
  1186.  
  1187. [Kmeetmecall-01*CLI>
  1188. [May 17 20:05:04] NOTICE[3375]: pbx_spool.c:366 attempt_thread: Call completed to SIP/0621899999@0852100340
  1189.  
  1190. [Kmeetmecall-01*CLI>
  1191. Really destroying SIP dialog '[email protected]' Method: BYE
  1192.  
  1193. [Kmeetmecall-01*CLI>
  1194. [May 17 20:05:07] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for [email protected]
  1195. REGISTER 12 headers, 0 lines
  1196. REGISTER attempt 1 to [email protected]
  1197. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  1198. REGISTER sip:itsp2.net:6060 SIP/2.0
  1199.  
  1200. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK1a344c59;rport
  1201.  
  1202. From: <sip:[email protected]>;tag=as3842065a
  1203.  
  1204. To: <sip:[email protected]>
  1205.  
  1206. Call-ID: [email protected]
  1207.  
  1208. CSeq: 171 REGISTER
  1209.  
  1210.  
  1211. [Kmeetmecall-01*CLI>
  1212. User-Agent: Asterisk PBX
  1213.  
  1214. Max-Forwards: 70
  1215.  
  1216. Expires: 120
  1217.  
  1218. Contact: <sip:[email protected]>
  1219.  
  1220. Event: registration
  1221.  
  1222. Content-Length: 0
  1223.  
  1224.  
  1225.  
  1226.  
  1227. ---
  1228.  
  1229. [Kmeetmecall-01*CLI>
  1230.  
  1231. <--- SIP read from xx.xxx.119.38:6060 --->
  1232. SIP/2.0 200 OK
  1233.  
  1234. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK1a344c59;rport
  1235.  
  1236. From: <sip:[email protected]>;tag=as3842065a
  1237.  
  1238. To: <sip:[email protected]>
  1239.  
  1240. Call-ID: [email protected]
  1241.  
  1242. CSeq: 171 REGISTER
  1243.  
  1244. Contact: <sip:[email protected]>
  1245.  
  1246. User-Agent: Asterisk PBX
  1247.  
  1248. Expires: 60
  1249.  
  1250. Event: registration
  1251.  
  1252. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  1253.  
  1254. Content-Length: 0
  1255.  
  1256.  
  1257.  
  1258.  
  1259. <------------->
  1260. --- (12 headers 0 lines) ---
  1261. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  1262. [May 17 20:05:07] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
  1263.  
  1264. [Kmeetmecall-01*CLI>
  1265. [May 17 20:05:07] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for [email protected]
  1266. REGISTER 12 headers, 0 lines
  1267. REGISTER attempt 1 to [email protected]
  1268. Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
  1269. REGISTER sip:itsp2.net:6060 SIP/2.0
  1270.  
  1271. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK75e2c0e0;rport
  1272.  
  1273. From: <sip:[email protected]>;tag=as777e7123
  1274.  
  1275. To: <sip:[email protected]>
  1276.  
  1277. Call-ID: [email protected]
  1278.  
  1279. CSeq: 171 REGISTER
  1280.  
  1281.  
  1282. [Kmeetmecall-01*CLI>
  1283. User-Agent: Asterisk PBX
  1284.  
  1285. Max-Forwards: 70
  1286.  
  1287. Expires: 120
  1288.  
  1289. Contact: <sip:[email protected]>
  1290.  
  1291. Event: registration
  1292.  
  1293. Content-Length: 0
  1294.  
  1295.  
  1296.  
  1297.  
  1298. ---
  1299.  
  1300. [Kmeetmecall-01*CLI>
  1301.  
  1302. <--- SIP read from xx.xxx.119.38:6060 --->
  1303. SIP/2.0 200 OK
  1304.  
  1305. Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK75e2c0e0;rport
  1306.  
  1307. From: <sip:[email protected]>;tag=as777e7123
  1308.  
  1309. To: <sip:[email protected]>
  1310.  
  1311. Call-ID: [email protected]
  1312.  
  1313. CSeq: 171 REGISTER
  1314.  
  1315. Contact: <sip:[email protected]>
  1316.  
  1317. User-Agent: Asterisk PBX
  1318.  
  1319. Expires: 60
  1320.  
  1321. Event: registration
  1322.  
  1323. Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
  1324.  
  1325. Content-Length: 0
  1326.  
  1327.  
  1328.  
  1329.  
  1330. <------------->
  1331. --- (12 headers 0 lines) ---
  1332. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  1333. [May 17 20:05:07] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
  1334.  
  1335. [Kmeetmecall-01*CLI>
  1336. Disconnected from Asterisk server
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