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Someone
Thursday, September 25th, 2008 at 2:06:00pm MDT 

  1. Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
  2. Created by Mark Spencer <markster@digium.com>
  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.   == Parsing '/etc/asterisk/asterisk.conf': Found
  9.   == Parsing '/etc/asterisk/extconfig.conf': Found
  10. Connected to Asterisk 1.4.21.2 currently running on pbx-b (pid = 4418)
  11. pbx-b*CLI>
  12. Verbosity is at least 3
  13.  
  14. [Kpbx-b*CLI>
  15. <--- SIP read from 192.168.50.250:5060 --->
  16. SIP/2.0 200 OK
  17. Call-ID: 54a2a33e79e3f7005605096920c7a334@192.168.50.1
  18. CSeq: 102 OPTIONS
  19. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as0372c1f1
  20. To: <sip:2620@192.168.50.250>;tag=5099e5a7560d7c4
  21. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK090939c6;rport
  22. Content-Length: 0
  23. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  24. Supported: replaces
  25. User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5
  26.  
  27.  
  28. <------------->
  29. --- (10 headers 0 lines) ---
  30.  
  31. [Kpbx-b*CLI>
  32. <--- SIP read from 192.168.50.250:5060 --->
  33. SIP/2.0 200 OK
  34. Call-ID: 49b233df4f41a6310b6bbd4f13e9df90@192.168.50.1
  35. CSeq: 102 OPTIONS
  36. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as30c562c4
  37. To: <sip:2606@192.168.50.250>;tag=68e3bd1fdbe51d0
  38. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6819ddc1;rport
  39. Content-Length: 0
  40. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  41. Supported: replaces
  42. User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5
  43.  
  44.  
  45. <------------->
  46. --- (10 headers 0 lines) ---
  47.  
  48. [Kpbx-b*CLI>
  49. Really destroying SIP dialog '49b233df4f41a6310b6bbd4f13e9df90@192.168.50.1' Method: OPTIONS
  50.  
  51. [Kpbx-b*CLI>
  52. Really destroying SIP dialog '54a2a33e79e3f7005605096920c7a334@192.168.50.1' Method: OPTIONS
  53.  
  54. [Kpbx-b*CLI>
  55. <--- SIP read from 192.168.50.249:5060 --->
  56. INVITE sip:2424@192.168.50.1:5060;user=phone SIP/2.0
  57. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F
  58. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  59. To: <sip:2424@192.168.50.1;user=phone>
  60. CSeq: 1 INVITE
  61. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  62. Contact: <sip:2608@192.168.50.249>
  63. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  64. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  65. Supported: 100rel,replaces
  66. Allow-Events: talk,hold,conference
  67. Max-Forwards: 70
  68. Content-Type: application/sdp
  69. Content-Length: 277
  70.  
  71. v=0
  72. o=- 1222371374 1222371374 IN IP4 192.168.50.249
  73. s=Polycom IP Phone
  74. c=IN IP4 192.168.50.249
  75. t=0 0
  76. m=audio 2234 RTP/AVP 9 0 8 18 101
  77. a=sendrecv
  78. a=rtpmap:9 G722/8000
  79. a=rtpmap:0 PCMU/8000
  80. a=rtpmap:8 PCMA/8000
  81. a=rtpmap:18 G729/8000
  82. a=rtpmap:101 telephone-event/8000
  83.  
  84. <------------->
  85. --- (14 headers 12 lines) ---
  86. Sending to 192.168.50.249 : 5060 (no NAT)
  87. Using INVITE request as basis request - 6c7b529a-ba914749-fe3faecc@192.168.50.249
  88.  
  89. <--- Reliably Transmitting (no NAT) to 192.168.50.249:5060 --->
  90. SIP/2.0 407 Proxy Authentication Required
  91. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F;received=192.168.50.249
  92. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  93. To: <sip:2424@192.168.50.1;user=phone>;tag=as35c92802
  94. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  95. CSeq: 1 INVITE
  96. User-Agent: Asterisk PBX
  97. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  98. Supported: replaces
  99. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09081313"
  100. Content-Length: 0
  101.  
  102.  
  103. <------------>
  104. Scheduling destruction of SIP dialog '6c7b529a-ba914749-fe3faecc@192.168.50.249' in 32000 ms (Method: INVITE)
  105. Found user '2608'
  106.  
  107. <--- SIP read from 192.168.50.249:5060 --->
  108. ACK sip:2424@192.168.50.1:5060 SIP/2.0
  109. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F
  110. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  111. To: <sip:2424@192.168.50.1;user=phone>;tag=as35c92802
  112. CSeq: 1 ACK
  113. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  114. Contact: <sip:2608@192.168.50.249>
  115. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  116. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  117. Max-Forwards: 70
  118. Content-Length: 0
  119.  
  120.  
  121. <------------->
  122. --- (11 headers 0 lines) ---
  123.  
  124. [Kpbx-b*CLI>
  125. <--- SIP read from 192.168.50.249:5060 --->
  126. INVITE sip:2424@192.168.50.1:5060;user=phone SIP/2.0
  127. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0
  128. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  129. To: <sip:2424@192.168.50.1;user=phone>
  130. CSeq: 2 INVITE
  131. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  132. Contact: <sip:2608@192.168.50.249>
  133. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  134. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  135. Supported: 100rel,replaces
  136. Allow-Events: talk,hold,conference
  137. Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="df84c8550a18f85d5aad62024ea0bfdb", algorithm=MD5
  138. Max-Forwards: 70
  139. Content-Type: application/sdp
  140. Content-Length: 277
  141.  
  142. v=0
  143. o=- 1222371374 1222371374 IN IP4 192.168.50.249
  144. s=Polycom IP Phone
  145. c=IN IP4 192.168.50.249
  146. t=0 0
  147. m=audio 2234 RTP/AVP 9 0 8 18 101
  148. a=sendrecv
  149. a=rtpmap:9 G722/8000
  150. a=rtpmap:0 PCMU/8000
  151. a=rtpmap:8 PCMA/8000
  152. a=rtpmap:18 G729/8000
  153. a=rtpmap:101 telephone-event/8000
  154.  
  155. <------------->
  156. --- (15 headers 12 lines) ---
  157. Sending to 192.168.50.249 : 5060 (no NAT)
  158. Using INVITE request as basis request - 6c7b529a-ba914749-fe3faecc@192.168.50.249
  159. Found user '2608'
  160. Found RTP audio format 9
  161. Found RTP audio format 0
  162. Found RTP audio format 8
  163. Found RTP audio format 18
  164. Found RTP audio format 101
  165. Peer audio RTP is at port 192.168.50.249:2234
  166. Found audio description format G722 for ID 9
  167. Found audio description format PCMU for ID 0
  168. Found audio description format PCMA for ID 8
  169. Found audio description format G729 for ID 18
  170. Found audio description format telephone-event for ID 101
  171. Capabilities: us - 0xc2e (gsm|ulaw|alaw|g726|adpcm|ilbc), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
  172. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  173. Peer audio RTP is at port 192.168.50.249:2234
  174. Looking for 2424 in standard (domain 192.168.50.1)
  175. list_route: hop: <sip:2608@192.168.50.249>
  176.  
  177. <--- Transmitting (no NAT) to 192.168.50.249:5060 --->
  178. SIP/2.0 100 Trying
  179. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249
  180. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  181. To: <sip:2424@192.168.50.1;user=phone>
  182. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  183. CSeq: 2 INVITE
  184. User-Agent: Asterisk PBX
  185. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  186. Supported: replaces
  187. Contact: <sip:2424@192.168.50.1>
  188. Content-Length: 0
  189.  
  190.  
  191. <------------>
  192.     -- Executing [2424@standard:1] Queue("SIP/2608-b7d05e10", "group1") in new stack
  193.  
  194. [Kpbx-b*CLI>
  195. Audio is at 192.168.50.1 port 16828
  196. Adding codec 0x4 (ulaw) to SDP
  197. Adding codec 0x8 (alaw) to SDP
  198. Adding non-codec 0x1 (telephone-event) to SDP
  199.  
  200. <--- Transmitting (no NAT) to 192.168.50.249:5060 --->
  201. SIP/2.0 183 Session Progress
  202. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249
  203. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  204. To: <sip:2424@192.168.50.1;user=phone>;tag=as2c27759c
  205. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  206. CSeq: 2 INVITE
  207. User-Agent: Asterisk PBX
  208. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  209. Supported: replaces
  210. Contact: <sip:2424@192.168.50.1>
  211. Content-Type: application/sdp
  212. Content-Length: 262
  213.  
  214. v=0
  215. o=root 4418 4418 IN IP4 192.168.50.1
  216. s=session
  217. c=IN IP4 192.168.50.1
  218. t=0 0
  219. m=audio 16828 RTP/AVP 0 8 101
  220. a=rtpmap:0 PCMU/8000
  221. a=rtpmap:8 PCMA/8000
  222. a=rtpmap:101 telephone-event/8000
  223. a=fmtp:101 0-16
  224. a=silenceSupp:off - - - -
  225. a=ptime:20
  226. a=sendrecv
  227.  
  228. <------------>
  229.     -- Started music on hold, class 'ring', on SIP/2608-b7d05e10
  230.     -- Executing [2604@standard:1] Set("Local/2604@standard-f909,2", "ARGS=dialExt,2604,no") in new stack
  231.     -- Executing [2604@standard:2] AGI("Local/2604@standard-f909,2", "callRoute3.agi") in new stack
  232.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  233.     -- Executing [2605@standard:1] Set("Local/2605@standard-4cfe,2", "ARGS=dialExt,2605,no") in new stack
  234.  
  235. [Kpbx-b*CLI>
  236.     -- Executing [2605@standard:2] AGI("Local/2605@standard-4cfe,2", "callRoute3.agi") in new stack
  237.  
  238. [Kpbx-b*CLI>
  239.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  240.     -- AGI Script callRoute3.agi completed, returning 0
  241.  
  242. [Kpbx-b*CLI>
  243.     -- Executing [2604@smartRing:1] Set("Local/2604@standard-f909,2", "phase=0") in new stack
  244.  
  245. [Kpbx-b*CLI>
  246.     -- Executing [2604@smartRing:2] Set("Local/2604@standard-f909,2", "phones=IAX2/2604") in new stack
  247.  
  248. [Kpbx-b*CLI>
  249.     -- Executing [2604@smartRing:3] Set("Local/2604@standard-f909,2", "duration=60") in new stack
  250.  
  251. [Kpbx-b*CLI>
  252.     -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-f909,2", "ALERT-INFO: null") in new stack
  253.     -- Executing [2604@smartRing:5] Dial("Local/2604@standard-f909,2", "IAX2/2604|60|rwWA()") in new stack
  254.     -- Called 2604
  255.  
  256. [Kpbx-b*CLI>
  257.     -- Local/2604@standard-f909,1 is ringing
  258.  
  259. [Kpbx-b*CLI>
  260.     -- Call accepted by 192.168.1.120 (format ulaw)
  261.  
  262. [Kpbx-b*CLI>
  263.     -- Format for call is ulaw
  264.  
  265. [Kpbx-b*CLI>
  266.     -- IAX2/2604-14650 is ringing
  267.  
  268. [Kpbx-b*CLI>
  269.     -- AGI Script callRoute3.agi completed, returning 0
  270.  
  271. [Kpbx-b*CLI>
  272.     -- Executing [2605@smartRing:1] Set("Local/2605@standard-4cfe,2", "phase=0") in new stack
  273.  
  274. [Kpbx-b*CLI>
  275.     -- Executing [2605@smartRing:2] Set("Local/2605@standard-4cfe,2", "phones=IAX2/2605") in new stack
  276.  
  277. [Kpbx-b*CLI>
  278.     -- Executing [2605@smartRing:3] Set("Local/2605@standard-4cfe,2", "duration=60") in new stack
  279.     -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-4cfe,2", "ALERT-INFO: null") in new stack
  280.     -- Executing [2605@smartRing:5] Dial("Local/2605@standard-4cfe,2", "IAX2/2605|60|rwWA()") in new stack
  281.     -- Called 2605
  282.  
  283. [Kpbx-b*CLI>
  284.     -- Local/2605@standard-4cfe,1 is ringing
  285.  
  286. [Kpbx-b*CLI>
  287.     -- Call accepted by 192.168.1.116 (format ulaw)
  288.     -- Format for call is ulaw
  289.     -- IAX2/2605-11114 is ringing
  290.  
  291. [Kpbx-b*CLI>
  292. <--- SIP read from 127.0.0.1:50603 --->
  293. REGISTER sip:127.0.0.1 SIP/2.0
  294. Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76
  295. Max-Forwards: 70
  296. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  297. To: <sip:SGATE@127.0.0.1>
  298. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  299. CSeq: 78407 REGISTER
  300. User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
  301. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
  302. Expires: 5
  303. Content-Length:  0
  304.  
  305.  
  306. <------------->
  307. --- (11 headers 0 lines) ---
  308. Using latest REGISTER request as basis request
  309. Sending to 127.0.0.1 : 50603 (NAT)
  310.  
  311. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  312. SIP/2.0 100 Trying
  313. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76;received=127.0.0.1;rport=50603
  314. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  315. To: <sip:SGATE@127.0.0.1>
  316. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  317. CSeq: 78407 REGISTER
  318. User-Agent: Asterisk PBX
  319. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  320. Supported: replaces
  321. Contact: <sip:SGATE@127.0.0.1>
  322. Content-Length: 0
  323.  
  324.  
  325. <------------>
  326.  
  327. [Kpbx-b*CLI>
  328. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  329. SIP/2.0 401 Unauthorized
  330. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76;received=127.0.0.1;rport=50603
  331. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  332. To: <sip:SGATE@127.0.0.1>;tag=as7a50f888
  333. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  334. CSeq: 78407 REGISTER
  335. User-Agent: Asterisk PBX
  336. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  337. Supported: replaces
  338. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ac1fe83"
  339. Content-Length: 0
  340.  
  341.  
  342. <------------>
  343.  
  344. [Kpbx-b*CLI>
  345. Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER)
  346.  
  347. <--- SIP read from 127.0.0.1:50603 --->
  348. REGISTER sip:127.0.0.1 SIP/2.0
  349. Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878
  350. Max-Forwards: 70
  351. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  352. To: <sip:SGATE@127.0.0.1>
  353. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  354. CSeq: 78408 REGISTER
  355. User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
  356. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
  357. Expires: 5
  358. Authorization: Digest username="SGATE", realm="asterisk", nonce="6ac1fe83", uri="sip:127.0.0.1", response="3352bd3a3d085ef384d6b4c0625745cf", algorithm=MD5
  359. Content-Length:  0
  360.  
  361.  
  362. <------------->
  363. --- (12 headers 0 lines) ---
  364. Using latest REGISTER request as basis request
  365. Sending to 127.0.0.1 : 50603 (NAT)
  366.  
  367. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  368. SIP/2.0 100 Trying
  369. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878;received=127.0.0.1;rport=50603
  370. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  371. To: <sip:SGATE@127.0.0.1>
  372. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  373. CSeq: 78408 REGISTER
  374. User-Agent: Asterisk PBX
  375. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  376. Supported: replaces
  377. Contact: <sip:SGATE@127.0.0.1>
  378. Content-Length: 0
  379.  
  380.  
  381. <------------>
  382.  
  383. [Kpbx-b*CLI>
  384. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  385. SIP/2.0 200 OK
  386. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878;received=127.0.0.1;rport=50603
  387. From: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
  388. To: <sip:SGATE@127.0.0.1>;tag=as7a50f888
  389. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  390. CSeq: 78408 REGISTER
  391. User-Agent: Asterisk PBX
  392. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  393. Supported: replaces
  394. Expires: 60
  395. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>;expires=60
  396. Date: Thu, 25 Sep 2008 20:01:50 GMT
  397. Content-Length: 0
  398.  
  399.  
  400. <------------>
  401. Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER)
  402.  
  403. [Kpbx-b*CLI>
  404.     -- Nobody picked up in 5000 ms
  405.     -- Nobody picked up in 5000 ms
  406.     -- Hungup 'IAX2/2604-14650'
  407.     -- Hungup 'IAX2/2605-11114'
  408.   == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-f909,2'
  409.  
  410. [Kpbx-b*CLI>
  411.   == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-4cfe,2'
  412.  
  413. [Kpbx-b*CLI>
  414. Really destroying SIP dialog 'b436e144-f17f4bab-a0a95a86@192.168.50.249' Method: ACK
  415.  
  416. [Kpbx-b*CLI>
  417.     -- Executing [2604@standard:1] Set("Local/2604@standard-b3cf,2", "ARGS=dialExt,2604,no") in new stack
  418.  
  419. [Kpbx-b*CLI>
  420.     -- Executing [2605@standard:1] Set("Local/2605@standard-ac07,2", "ARGS=dialExt,2605,no") in new stack
  421.     -- Executing [2605@standard:2] AGI("Local/2605@standard-ac07,2", "callRoute3.agi") in new stack
  422.     -- Executing [2604@standard:2] AGI("Local/2604@standard-b3cf,2", "callRoute3.agi") in new stack
  423.  
  424. [Kpbx-b*CLI>
  425.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  426.  
  427. [Kpbx-b*CLI>
  428.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  429.     -- AGI Script callRoute3.agi completed, returning 0
  430.  
  431. [Kpbx-b*CLI>
  432.     -- Executing [2605@smartRing:1] Set("Local/2605@standard-ac07,2", "phase=0") in new stack
  433.  
  434. [Kpbx-b*CLI>
  435.     -- Executing [2605@smartRing:2] Set("Local/2605@standard-ac07,2", "phones=IAX2/2605") in new stack
  436.  
  437. [Kpbx-b*CLI>
  438.     -- Executing [2605@smartRing:3] Set("Local/2605@standard-ac07,2", "duration=60") in new stack
  439.     -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-ac07,2", "ALERT-INFO: null") in new stack
  440.     -- Executing [2605@smartRing:5] Dial("Local/2605@standard-ac07,2", "IAX2/2605|60|rwWA()") in new stack
  441.     -- Called 2605
  442.  
  443. [Kpbx-b*CLI>
  444.     -- Call accepted by 192.168.1.116 (format ulaw)
  445.     -- Format for call is ulaw
  446.     -- Local/2605@standard-ac07,1 is ringing
  447.  
  448. [Kpbx-b*CLI>
  449.     -- IAX2/2605-3281 is ringing
  450.     -- AGI Script callRoute3.agi completed, returning 0
  451.     -- Executing [2604@smartRing:1] Set("Local/2604@standard-b3cf,2", "phase=0") in new stack
  452.  
  453. [Kpbx-b*CLI>
  454.     -- Executing [2604@smartRing:2] Set("Local/2604@standard-b3cf,2", "phones=IAX2/2604") in new stack
  455.  
  456. [Kpbx-b*CLI>
  457.     -- Executing [2604@smartRing:3] Set("Local/2604@standard-b3cf,2", "duration=60") in new stack
  458.  
  459. [Kpbx-b*CLI>
  460.     -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-b3cf,2", "ALERT-INFO: null") in new stack
  461.     -- Executing [2604@smartRing:5] Dial("Local/2604@standard-b3cf,2", "IAX2/2604|60|rwWA()") in new stack
  462.     -- Called 2604
  463.  
  464. [Kpbx-b*CLI>
  465.     -- Local/2604@standard-b3cf,1 is ringing
  466.  
  467. [Kpbx-b*CLI>
  468.     -- Call accepted by 192.168.1.120 (format ulaw)
  469.     -- Format for call is ulaw
  470.     -- IAX2/2604-9064 is ringing
  471.  
  472. [Kpbx-b*CLI>
  473.     -- Remote UNIX connection
  474.  
  475. [Kpbx-b*CLI>
  476.     -- Nobody picked up in 5000 ms
  477.  
  478. [Kpbx-b*CLI>
  479.     -- Nobody picked up in 5000 ms
  480.  
  481. [Kpbx-b*CLI>
  482.     -- Hungup 'IAX2/2604-9064'
  483.     -- Hungup 'IAX2/2605-3281'
  484.   == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-ac07,2'
  485.   == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-b3cf,2'
  486.     -- Remote UNIX connection disconnected
  487.  
  488. [Kpbx-b*CLI>
  489.     -- Remote UNIX connection
  490.  
  491. [Kpbx-b*CLI>
  492.     -- Remote UNIX connection disconnected
  493.  
  494. [Kpbx-b*CLI>
  495.     -- Remote UNIX connection
  496.  
  497. [Kpbx-b*CLI>
  498.     -- Remote UNIX connection disconnected
  499.  
  500. [Kpbx-b*CLI>
  501.     -- Remote UNIX connection
  502.  
  503. [Kpbx-b*CLI>
  504.     -- Remote UNIX connection disconnected
  505.  
  506. [Kpbx-b*CLI>
  507.     -- Remote UNIX connection
  508.  
  509. [Kpbx-b*CLI>
  510.     -- Remote UNIX connection disconnected
  511.  
  512. [Kpbx-b*CLI>
  513.   == Parsing '/etc/asterisk/manager.conf': Found
  514.  
  515. [Kpbx-b*CLI>
  516.   == Manager 'admin' logged on from 127.0.0.1
  517.  
  518. [Kpbx-b*CLI>
  519.   == Manager 'admin' logged off from 127.0.0.1
  520.  
  521. [Kpbx-b*CLI>
  522.     -- Stopped music on hold on SIP/2608-b7d05e10
  523.     -- Playing periodic announcement
  524.     -- <SIP/2608-b7d05e10> Playing 'queue-periodic-announce' (language 'en')
  525.     -- Started music on hold, class 'ring', on SIP/2608-b7d05e10
  526.  
  527. [Kpbx-b*CLI>
  528.     -- Executing [2604@standard:1] Set("Local/2604@standard-34d8,2", "ARGS=dialExt,2604,no") in new stack
  529.  
  530. [Kpbx-b*CLI>
  531.     -- Executing [2604@standard:2] AGI("Local/2604@standard-34d8,2", "callRoute3.agi") in new stack
  532.  
  533. [Kpbx-b*CLI>
  534.     -- Executing [2605@standard:1] Set("Local/2605@standard-583e,2", "ARGS=dialExt,2605,no") in new stack
  535.  
  536. [Kpbx-b*CLI>
  537.     -- Executing [2605@standard:2] AGI("Local/2605@standard-583e,2", "callRoute3.agi") in new stack
  538.  
  539. [Kpbx-b*CLI>
  540.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  541.  
  542. [Kpbx-b*CLI>
  543.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  544.     -- AGI Script callRoute3.agi completed, returning 0
  545.  
  546. [Kpbx-b*CLI>
  547.     -- Executing [2605@smartRing:1] Set("Local/2605@standard-583e,2", "phase=0") in new stack
  548.  
  549. [Kpbx-b*CLI>
  550.     -- Executing [2605@smartRing:2] Set("Local/2605@standard-583e,2", "phones=IAX2/2605") in new stack
  551.  
  552. [Kpbx-b*CLI>
  553.     -- Executing [2605@smartRing:3] Set("Local/2605@standard-583e,2", "duration=60") in new stack
  554.     -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-583e,2", "ALERT-INFO: null") in new stack
  555.     -- Executing [2605@smartRing:5] Dial("Local/2605@standard-583e,2", "IAX2/2605|60|rwWA()") in new stack
  556.  
  557. [Kpbx-b*CLI>
  558.     -- Called 2605
  559.  
  560. [Kpbx-b*CLI>
  561.     -- Local/2605@standard-583e,1 is ringing
  562.  
  563. [Kpbx-b*CLI>
  564.     -- Call accepted by 192.168.1.116 (format ulaw)
  565.     -- Format for call is ulaw
  566.     -- IAX2/2605-8300 is ringing
  567.     -- AGI Script callRoute3.agi completed, returning 0
  568.  
  569. [Kpbx-b*CLI>
  570.     -- Executing [2604@smartRing:1] Set("Local/2604@standard-34d8,2", "phase=0") in new stack
  571.  
  572. [Kpbx-b*CLI>
  573.     -- Executing [2604@smartRing:2] Set("Local/2604@standard-34d8,2", "phones=IAX2/2604") in new stack
  574.  
  575. [Kpbx-b*CLI>
  576.     -- Executing [2604@smartRing:3] Set("Local/2604@standard-34d8,2", "duration=60") in new stack
  577.     -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-34d8,2", "ALERT-INFO: null") in new stack
  578.     -- Executing [2604@smartRing:5] Dial("Local/2604@standard-34d8,2", "IAX2/2604|60|rwWA()") in new stack
  579.     -- Called 2604
  580.  
  581. [Kpbx-b*CLI>
  582.     -- Local/2604@standard-34d8,1 is ringing
  583.  
  584. [Kpbx-b*CLI>
  585.     -- Call accepted by 192.168.1.120 (format ulaw)
  586.     -- Format for call is ulaw
  587.     -- IAX2/2604-6615 is ringing
  588.  
  589. [Kpbx-b*CLI>
  590.     -- Nobody picked up in 5000 ms
  591.  
  592. [Kpbx-b*CLI>
  593.     -- Nobody picked up in 5000 ms
  594.  
  595. [Kpbx-b*CLI>
  596.     -- Hungup 'IAX2/2604-6615'
  597.  
  598. [Kpbx-b*CLI>
  599.     -- Hungup 'IAX2/2605-8300'
  600.   == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-583e,2'
  601.   == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-34d8,2'
  602.  
  603. [Kpbx-b*CLI>
  604. Really destroying SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' Method: REGISTER
  605.  
  606. [Kpbx-b*CLI>
  607.     -- Stopped music on hold on SIP/2608-b7d05e10
  608.     -- Playing periodic announcement
  609.     -- <SIP/2608-b7d05e10> Playing 'queue-periodic-announce' (language 'en')
  610.     -- Started music on hold, class 'ring', on SIP/2608-b7d05e10
  611.  
  612. [Kpbx-b*CLI>
  613.     -- Executing [2604@standard:1] Set("Local/2604@standard-1f6c,2", "ARGS=dialExt,2604,no") in new stack
  614.  
  615. [Kpbx-b*CLI>
  616.     -- Executing [2604@standard:2] AGI("Local/2604@standard-1f6c,2", "callRoute3.agi") in new stack
  617.  
  618. [Kpbx-b*CLI>
  619.     -- Executing [2605@standard:1] Set("Local/2605@standard-a9b5,2", "ARGS=dialExt,2605,no") in new stack
  620.  
  621. [Kpbx-b*CLI>
  622.     -- Executing [2605@standard:2] AGI("Local/2605@standard-a9b5,2", "callRoute3.agi") in new stack
  623.  
  624. [Kpbx-b*CLI>
  625.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  626.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  627.     -- AGI Script callRoute3.agi completed, returning 0
  628.  
  629. [Kpbx-b*CLI>
  630.     -- Executing [2604@smartRing:1] Set("Local/2604@standard-1f6c,2", "phase=0") in new stack
  631.  
  632. [Kpbx-b*CLI>
  633.     -- Executing [2604@smartRing:2] Set("Local/2604@standard-1f6c,2", "phones=IAX2/2604") in new stack
  634.  
  635. [Kpbx-b*CLI>
  636.     -- Executing [2604@smartRing:3] Set("Local/2604@standard-1f6c,2", "duration=60") in new stack
  637.     -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-1f6c,2", "ALERT-INFO: null") in new stack
  638.     -- Executing [2604@smartRing:5] Dial("Local/2604@standard-1f6c,2", "IAX2/2604|60|rwWA()") in new stack
  639.  
  640. [Kpbx-b*CLI>
  641.     -- Called 2604
  642.  
  643. [Kpbx-b*CLI>
  644.     -- Local/2604@standard-1f6c,1 is ringing
  645.  
  646. [Kpbx-b*CLI>
  647.     -- Call accepted by 192.168.1.120 (format ulaw)
  648.     -- Format for call is ulaw
  649.     -- IAX2/2604-10243 is ringing
  650.     -- AGI Script callRoute3.agi completed, returning 0
  651.  
  652. [Kpbx-b*CLI>
  653.     -- Executing [2605@smartRing:1] Set("Local/2605@standard-a9b5,2", "phase=0") in new stack
  654.  
  655. [Kpbx-b*CLI>
  656.     -- Executing [2605@smartRing:2] Set("Local/2605@standard-a9b5,2", "phones=IAX2/2605") in new stack
  657.  
  658. [Kpbx-b*CLI>
  659.     -- Executing [2605@smartRing:3] Set("Local/2605@standard-a9b5,2", "duration=60") in new stack
  660.     -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-a9b5,2", "ALERT-INFO: null") in new stack
  661.     -- Executing [2605@smartRing:5] Dial("Local/2605@standard-a9b5,2", "IAX2/2605|60|rwWA()") in new stack
  662.  
  663. [Kpbx-b*CLI>
  664.     -- Called 2605
  665.  
  666. [Kpbx-b*CLI>
  667.     -- Local/2605@standard-a9b5,1 is ringing
  668.  
  669. [Kpbx-b*CLI>
  670.     -- Call accepted by 192.168.1.116 (format ulaw)
  671.     -- Format for call is ulaw
  672.     -- IAX2/2605-6419 is ringing
  673.  
  674. [Kpbx-b*CLI>
  675.     -- Nobody picked up in 5000 ms
  676.     -- Nobody picked up in 5000 ms
  677.     -- Hungup 'IAX2/2605-6419'
  678.     -- Hungup 'IAX2/2604-10243'
  679.   == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-a9b5,2'
  680.   == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-1f6c,2'
  681. Reliably Transmitting (no NAT) to 192.168.50.249:5060:
  682. OPTIONS sip:2625@192.168.50.249 SIP/2.0
  683. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK303f90c1;rport
  684. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as5108246a
  685. To: <sip:2625@192.168.50.249>
  686. Contact: <sip:asterisk@192.168.50.1>
  687. Call-ID: 7a3aee0f48e291831d8534c117692bc1@192.168.50.1
  688. CSeq: 102 OPTIONS
  689. User-Agent: Asterisk PBX
  690. Max-Forwards: 70
  691. Date: Thu, 25 Sep 2008 20:02:39 GMT
  692. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  693. Supported: replaces
  694. Content-Length: 0
  695.  
  696.  
  697. ---
  698.  
  699. [Kpbx-b*CLI>
  700. <--- SIP read from 192.168.50.249:5060 --->
  701. SIP/2.0 200 OK
  702. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK303f90c1;rport
  703. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as5108246a
  704. To: <sip:2625@192.168.50.249>;tag=25D2A55B-7CAD5A76
  705. CSeq: 102 OPTIONS
  706. Call-ID: 7a3aee0f48e291831d8534c117692bc1@192.168.50.1
  707. Contact: <sip:2625@192.168.50.249>
  708. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  709. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  710. Content-Length: 0
  711.  
  712.  
  713. <------------->
  714. --- (10 headers 0 lines) ---
  715.  
  716. [Kpbx-b*CLI>
  717. Really destroying SIP dialog '7a3aee0f48e291831d8534c117692bc1@192.168.50.1' Method: OPTIONS
  718.  
  719. [Kpbx-b*CLI>
  720. Reliably Transmitting (no NAT) to 192.168.50.249:5060:
  721. OPTIONS sip:2621@192.168.50.249 SIP/2.0
  722. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK31e2eb5b;rport
  723. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as784262cc
  724. To: <sip:2621@192.168.50.249>
  725. Contact: <sip:asterisk@192.168.50.1>
  726. Call-ID: 3cdd404a1b779ba56529cf84346b75b5@192.168.50.1
  727. CSeq: 102 OPTIONS
  728. User-Agent: Asterisk PBX
  729. Max-Forwards: 70
  730. Date: Thu, 25 Sep 2008 20:02:39 GMT
  731. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  732. Supported: replaces
  733. Content-Length: 0
  734.  
  735.  
  736. ---
  737.  
  738. [Kpbx-b*CLI>
  739. <--- SIP read from 192.168.50.249:5060 --->
  740. SIP/2.0 200 OK
  741. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK31e2eb5b;rport
  742. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as784262cc
  743. To: <sip:2621@192.168.50.249>;tag=4CB55808-105D791F
  744. CSeq: 102 OPTIONS
  745. Call-ID: 3cdd404a1b779ba56529cf84346b75b5@192.168.50.1
  746. Contact: <sip:2621@192.168.50.249>
  747. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  748. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  749. Content-Length: 0
  750.  
  751.  
  752. <------------->
  753. --- (10 headers 0 lines) ---
  754.  
  755. [Kpbx-b*CLI>
  756. Really destroying SIP dialog '3cdd404a1b779ba56529cf84346b75b5@192.168.50.1' Method: OPTIONS
  757.  
  758. [Kpbx-b*CLI>
  759. Reliably Transmitting (no NAT) to 192.168.50.249:5060:
  760. OPTIONS sip:2608@192.168.50.249 SIP/2.0
  761. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK3a984c2e;rport
  762. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as61c401bc
  763. To: <sip:2608@192.168.50.249>
  764. Contact: <sip:asterisk@192.168.50.1>
  765. Call-ID: 734a085e7c521b71339c75533595398e@192.168.50.1
  766. CSeq: 102 OPTIONS
  767. User-Agent: Asterisk PBX
  768. Max-Forwards: 70
  769. Date: Thu, 25 Sep 2008 20:02:39 GMT
  770. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  771. Supported: replaces
  772. Content-Length: 0
  773.  
  774.  
  775. ---
  776.  
  777. [Kpbx-b*CLI>
  778. <--- SIP read from 192.168.50.249:5060 --->
  779. SIP/2.0 200 OK
  780. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK3a984c2e;rport
  781. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as61c401bc
  782. To: <sip:2608@192.168.50.249>;tag=D2904F59-CD778D9C
  783. CSeq: 102 OPTIONS
  784. Call-ID: 734a085e7c521b71339c75533595398e@192.168.50.1
  785. Contact: <sip:2608@192.168.50.249>
  786. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  787. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  788. Content-Length: 0
  789.  
  790.  
  791. <------------->
  792. --- (10 headers 0 lines) ---
  793.  
  794. [Kpbx-b*CLI>
  795. Really destroying SIP dialog '734a085e7c521b71339c75533595398e@192.168.50.1' Method: OPTIONS
  796.  
  797. [Kpbx-b*CLI>
  798. Reliably Transmitting (no NAT) to 192.168.2.76:50603:
  799. OPTIONS sip:SGATE@192.168.2.76:50603;transport=UDP SIP/2.0
  800. Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2601acc9;rport
  801. From: "asterisk" <sip:asterisk@192.168.2.76>;tag=as299be91b
  802. To: <sip:SGATE@192.168.2.76:50603;transport=UDP>
  803. Contact: <sip:asterisk@192.168.2.76>
  804. Call-ID: 4f7b379049cc3019348e2e9c22816ac3@192.168.2.76
  805. CSeq: 102 OPTIONS
  806. User-Agent: Asterisk PBX
  807. Max-Forwards: 70
  808. Date: Thu, 25 Sep 2008 20:02:39 GMT
  809. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  810. Supported: replaces
  811. Content-Length: 0
  812.  
  813.  
  814. ---
  815.  
  816. <--- SIP read from 192.168.2.76:50603 --->
  817. SIP/2.0 200 OK
  818. Via: SIP/2.0/UDP 192.168.2.76:5060;rport=5060;received=192.168.2.76;branch=z9hG4bK2601acc9
  819. Call-ID: 4f7b379049cc3019348e2e9c22816ac3@192.168.2.76
  820. From: "asterisk" <sip:asterisk@192.168.2.76>;tag=as299be91b
  821. To: <sip:SGATE@192.168.2.76>
  822. CSeq: 102 OPTIONS
  823. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
  824. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
  825. Supported: replaces, 100rel, norefersub
  826. Allow-Events: presence, refer
  827. User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
  828. Content-Type: application/sdp
  829. Content-Length:   287
  830.  
  831. v=0
  832. o=- 3431361759 3431361759 IN IP4 192.168.2.76
  833. s=pjmedia
  834. c=IN IP4 192.168.2.76
  835. t=0 0
  836. m=audio 4000 RTP/AVP 3 0 8 101
  837. a=rtcp:4001 IN IP4 192.168.2.76
  838. a=rtpmap:3 GSM/8000
  839. a=rtpmap:0 PCMU/8000
  840. a=rtpmap:8 PCMA/8000
  841. a=sendrecv
  842. a=rtpmap:101 telephone-event/8000
  843. a=fmtp:101 0-15
  844.  
  845. <------------->
  846. --- (13 headers 13 lines) ---
  847.  
  848. [Kpbx-b*CLI>
  849. Really destroying SIP dialog '4f7b379049cc3019348e2e9c22816ac3@192.168.2.76' Method: OPTIONS
  850.  
  851. [Kpbx-b*CLI>
  852. Reliably Transmitting (no NAT) to 192.168.50.200:5060:
  853. OPTIONS sip:2601@192.168.50.200 SIP/2.0
  854. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6feadd47;rport
  855. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as6a633f8d
  856. To: <sip:2601@192.168.50.200>
  857. Contact: <sip:asterisk@192.168.50.1>
  858. Call-ID: 484e0e5330d466910934e57406ea086e@192.168.50.1
  859. CSeq: 102 OPTIONS
  860. User-Agent: Asterisk PBX
  861. Max-Forwards: 70
  862. Date: Thu, 25 Sep 2008 20:02:40 GMT
  863. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  864. Supported: replaces
  865. Content-Length: 0
  866.  
  867.  
  868. ---
  869.  
  870. [Kpbx-b*CLI>
  871. <--- SIP read from 192.168.50.200:5060 --->
  872. SIP/2.0 200 OK
  873. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6feadd47;rport
  874. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as6a633f8d
  875. To: <sip:2601@192.168.50.200>;tag=830DABD5-59DDE6F0
  876. CSeq: 102 OPTIONS
  877. Call-ID: 484e0e5330d466910934e57406ea086e@192.168.50.1
  878. Contact: <sip:2601@192.168.50.200>
  879. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  880. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.2.0084
  881. Content-Length: 0
  882.  
  883.  
  884. <------------->
  885. --- (10 headers 0 lines) ---
  886. Really destroying SIP dialog '484e0e5330d466910934e57406ea086e@192.168.50.1' Method: OPTIONS
  887. Reliably Transmitting (no NAT) to 192.168.50.245:5060:
  888. OPTIONS sip:2602@192.168.50.245 SIP/2.0
  889. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK1fe17f07;rport
  890. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as6689a6dd
  891. To: <sip:2602@192.168.50.245>
  892. Contact: <sip:asterisk@192.168.50.1>
  893. Call-ID: 74cfcd0c066a560c63bbfbf634c42060@192.168.50.1
  894. CSeq: 102 OPTIONS
  895. User-Agent: Asterisk PBX
  896. Max-Forwards: 70
  897. Date: Thu, 25 Sep 2008 20:02:41 GMT
  898. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  899. Supported: replaces
  900. Content-Length: 0
  901.  
  902.  
  903. ---
  904.  
  905. [Kpbx-b*CLI>
  906. <--- SIP read from 192.168.50.245:5060 --->
  907. SIP/2.0 200 OK
  908. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK1fe17f07;rport
  909. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as6689a6dd
  910. To: <sip:2602@192.168.50.245>;tag=86606091-5408CED2
  911. CSeq: 102 OPTIONS
  912. Call-ID: 74cfcd0c066a560c63bbfbf634c42060@192.168.50.1
  913. Contact: <sip:2602@192.168.50.245>
  914. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  915. User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084
  916. Content-Length: 0
  917.  
  918.  
  919. <------------->
  920. --- (10 headers 0 lines) ---
  921.  
  922. [Kpbx-b*CLI>
  923.     -- Stopped music on hold on SIP/2608-b7d05e10
  924.     -- Playing periodic announcement
  925.     -- <SIP/2608-b7d05e10> Playing 'queue-periodic-announce' (language 'en')
  926. Really destroying SIP dialog '74cfcd0c066a560c63bbfbf634c42060@192.168.50.1' Method: OPTIONS
  927.  
  928. [Kpbx-b*CLI>
  929. Reliably Transmitting (no NAT) to 192.168.50.247:5060:
  930. OPTIONS sip:2609@192.168.50.247:5060;transport=udp SIP/2.0
  931. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK5af09efc;rport
  932. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as4e3260c7
  933. To: <sip:2609@192.168.50.247:5060;transport=udp>
  934. Contact: <sip:asterisk@192.168.50.1>
  935. Call-ID: 285e067815cc3ba3240214021c4476ce@192.168.50.1
  936. CSeq: 102 OPTIONS
  937. User-Agent: Asterisk PBX
  938. Max-Forwards: 70
  939. Date: Thu, 25 Sep 2008 20:02:41 GMT
  940. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  941. Supported: replaces
  942. Content-Length: 0
  943.  
  944.  
  945. ---
  946.  
  947. [Kpbx-b*CLI>
  948. <--- SIP read from 192.168.50.247:5060 --->
  949. SIP/2.0 200 OK
  950. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK5af09efc;rport=5060;received=192.168.50.1
  951. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as4e3260c7
  952. To: <sip:2609@192.168.50.247:5060;transport=udp>;tag=1592297787
  953. Call-ID: 285e067815cc3ba3240214021c4476ce@192.168.50.1
  954. CSeq: 102 OPTIONS
  955. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  956. Server: Aastra 53i/2.1.0.2145
  957. Content-Length: 0
  958.  
  959.  
  960. <------------->
  961. --- (9 headers 0 lines) ---
  962.  
  963. [Kpbx-b*CLI>
  964. Really destroying SIP dialog '285e067815cc3ba3240214021c4476ce@192.168.50.1' Method: OPTIONS
  965.  
  966. [Kpbx-b*CLI>
  967. Reliably Transmitting (no NAT) to 192.168.1.220:5064:
  968. OPTIONS sip:2603@192.168.1.220:5064;transport=udp SIP/2.0
  969. Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK0c5a9343;rport
  970. From: "asterisk" <sip:asterisk@192.168.2.76>;tag=as69b07436
  971. To: <sip:2603@192.168.1.220:5064;transport=udp>
  972. Contact: <sip:asterisk@192.168.2.76>
  973. Call-ID: 00c5e755409d1a5a145e2cef76222eab@192.168.2.76
  974. CSeq: 102 OPTIONS
  975. User-Agent: Asterisk PBX
  976. Max-Forwards: 70
  977. Date: Thu, 25 Sep 2008 20:02:42 GMT
  978. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  979. Supported: replaces
  980. Content-Length: 0
  981.  
  982.  
  983. ---
  984.  
  985. [Kpbx-b*CLI>
  986. <--- SIP read from 192.168.1.220:5064 --->
  987. SIP/2.0 200 OK
  988. Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK0c5a9343;rport
  989. From: "asterisk" <sip:asterisk@192.168.2.76:5060>;tag=as69b07436
  990. To: <sip:2603@192.168.1.220:5064;transport=udp>;tag=8597659733de1f50
  991. Call-ID: 00c5e755409d1a5a145e2cef76222eab@192.168.2.76
  992. CSeq: 102 OPTIONS
  993. User-Agent: Grandstream GXP2000 1.1.5.15
  994. Contact: <sip:2603@192.168.1.220:5064;transport=udp>
  995. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  996. Supported: replaces, timer
  997. Content-Length: 0
  998.  
  999.  
  1000. <------------->
  1001. --- (11 headers 0 lines) ---
  1002.  
  1003. [Kpbx-b*CLI>
  1004. Really destroying SIP dialog '00c5e755409d1a5a145e2cef76222eab@192.168.2.76' Method: OPTIONS
  1005.  
  1006. [Kpbx-b*CLI>
  1007. Reliably Transmitting (no NAT) to 192.168.50.250:5060:
  1008. OPTIONS sip:2620@192.168.50.250 SIP/2.0
  1009. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK37e5b43f;rport
  1010. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as065d9faa
  1011. To: <sip:2620@192.168.50.250>
  1012. Contact: <sip:asterisk@192.168.50.1>
  1013. Call-ID: 3104ff051a5e5dc017d2d140416eba5b@192.168.50.1
  1014. CSeq: 102 OPTIONS
  1015. User-Agent: Asterisk PBX
  1016. Max-Forwards: 70
  1017. Date: Thu, 25 Sep 2008 20:02:44 GMT
  1018. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1019. Supported: replaces
  1020. Content-Length: 0
  1021.  
  1022.  
  1023. ---
  1024.  
  1025. [Kpbx-b*CLI>
  1026. Reliably Transmitting (no NAT) to 192.168.50.250:5060:
  1027. OPTIONS sip:2606@192.168.50.250 SIP/2.0
  1028. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK257e4467;rport
  1029. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as743f8842
  1030. To: <sip:2606@192.168.50.250>
  1031. Contact: <sip:asterisk@192.168.50.1>
  1032. Call-ID: 3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1
  1033. CSeq: 102 OPTIONS
  1034. User-Agent: Asterisk PBX
  1035. Max-Forwards: 70
  1036. Date: Thu, 25 Sep 2008 20:02:44 GMT
  1037. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1038. Supported: replaces
  1039. Content-Length: 0
  1040.  
  1041.  
  1042. ---
  1043.  
  1044. [Kpbx-b*CLI>
  1045. <--- SIP read from 192.168.50.250:5060 --->
  1046. SIP/2.0 200 OK
  1047. Call-ID: 3104ff051a5e5dc017d2d140416eba5b@192.168.50.1
  1048. CSeq: 102 OPTIONS
  1049. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as065d9faa
  1050. To: <sip:2620@192.168.50.250>;tag=32a8fe2eea25b02
  1051. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK37e5b43f;rport
  1052. Content-Length: 0
  1053. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  1054. Supported: replaces
  1055. User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5
  1056.  
  1057.  
  1058. <------------->
  1059. --- (10 headers 0 lines) ---
  1060.  
  1061. [Kpbx-b*CLI>
  1062. <--- SIP read from 192.168.50.250:5060 --->
  1063. SIP/2.0 200 OK
  1064. Call-ID: 3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1
  1065. CSeq: 102 OPTIONS
  1066. From: "asterisk" <sip:asterisk@192.168.50.1>;tag=as743f8842
  1067. To: <sip:2606@192.168.50.250>;tag=ad741e34d25cfb0
  1068. Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK257e4467;rport
  1069. Content-Length: 0
  1070. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  1071. Supported: replaces
  1072. User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5
  1073.  
  1074.  
  1075. <------------->
  1076. --- (10 headers 0 lines) ---
  1077.  
  1078. [Kpbx-b*CLI>
  1079. Really destroying SIP dialog '3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1' Method: OPTIONS
  1080.  
  1081. [Kpbx-b*CLI>
  1082. Really destroying SIP dialog '3104ff051a5e5dc017d2d140416eba5b@192.168.50.1' Method: OPTIONS
  1083.  
  1084. [Kpbx-b*CLI>
  1085. <--- SIP read from 127.0.0.1:50603 --->
  1086. REGISTER sip:127.0.0.1 SIP/2.0
  1087. Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764
  1088. Max-Forwards: 70
  1089. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1090. To: <sip:SGATE@127.0.0.1>
  1091. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1092. CSeq: 78409 REGISTER
  1093. User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
  1094. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
  1095. Expires: 5
  1096. Content-Length:  0
  1097.  
  1098.  
  1099. <------------->
  1100. --- (11 headers 0 lines) ---
  1101. Using latest REGISTER request as basis request
  1102. Sending to 127.0.0.1 : 50603 (NAT)
  1103.  
  1104. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  1105. SIP/2.0 100 Trying
  1106. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764;received=127.0.0.1;rport=50603
  1107. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1108. To: <sip:SGATE@127.0.0.1>
  1109. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1110. CSeq: 78409 REGISTER
  1111. User-Agent: Asterisk PBX
  1112. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1113. Supported: replaces
  1114. Contact: <sip:SGATE@127.0.0.1>
  1115. Content-Length: 0
  1116.  
  1117.  
  1118. <------------>
  1119.  
  1120. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  1121. SIP/2.0 401 Unauthorized
  1122. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764;received=127.0.0.1;rport=50603
  1123. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1124. To: <sip:SGATE@127.0.0.1>;tag=as040f4bd0
  1125. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1126. CSeq: 78409 REGISTER
  1127. User-Agent: Asterisk PBX
  1128. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1129. Supported: replaces
  1130. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ad6254b"
  1131. Content-Length: 0
  1132.  
  1133.  
  1134. <------------>
  1135. Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER)
  1136.  
  1137. [Kpbx-b*CLI>
  1138. <--- SIP read from 127.0.0.1:50603 --->
  1139. REGISTER sip:127.0.0.1 SIP/2.0
  1140. Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a
  1141. Max-Forwards: 70
  1142. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1143. To: <sip:SGATE@127.0.0.1>
  1144. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1145. CSeq: 78410 REGISTER
  1146. User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
  1147. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
  1148. Expires: 5
  1149. Authorization: Digest username="SGATE", realm="asterisk", nonce="3ad6254b", uri="sip:127.0.0.1", response="feacdd0352932ef92b607f7b7972f7d7", algorithm=MD5
  1150. Content-Length:  0
  1151.  
  1152.  
  1153. <------------->
  1154. --- (12 headers 0 lines) ---
  1155. Using latest REGISTER request as basis request
  1156. Sending to 127.0.0.1 : 50603 (NAT)
  1157.  
  1158. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  1159. SIP/2.0 100 Trying
  1160. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a;received=127.0.0.1;rport=50603
  1161. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1162. To: <sip:SGATE@127.0.0.1>
  1163. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1164. CSeq: 78410 REGISTER
  1165. User-Agent: Asterisk PBX
  1166. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1167. Supported: replaces
  1168. Contact: <sip:SGATE@127.0.0.1>
  1169. Content-Length: 0
  1170.  
  1171.  
  1172. <------------>
  1173.  
  1174. [Kpbx-b*CLI>
  1175. <--- Transmitting (no NAT) to 192.168.2.76:50603 --->
  1176. SIP/2.0 200 OK
  1177. Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a;received=127.0.0.1;rport=50603
  1178. From: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
  1179. To: <sip:SGATE@127.0.0.1>;tag=as040f4bd0
  1180. Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
  1181. CSeq: 78410 REGISTER
  1182. User-Agent: Asterisk PBX
  1183. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1184. Supported: replaces
  1185. Expires: 60
  1186. Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>;expires=60
  1187. Date: Thu, 25 Sep 2008 20:02:45 GMT
  1188. Content-Length: 0
  1189.  
  1190.  
  1191. <------------>
  1192.  
  1193. [Kpbx-b*CLI>
  1194. Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER)
  1195.  
  1196. [Kpbx-b*CLI>
  1197. <--- SIP read from 192.168.50.249:5060 --->
  1198. CANCEL sip:2424@192.168.50.1:5060;user=phone SIP/2.0
  1199. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0
  1200. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  1201. To: <sip:2424@192.168.50.1;user=phone>
  1202. CSeq: 2 CANCEL
  1203. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  1204. Contact: <sip:2608@192.168.50.249>
  1205. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  1206. Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="df84c8550a18f85d5aad62024ea0bfdb", algorithm=MD5
  1207. Max-Forwards: 70
  1208. Content-Length: 0
  1209.  
  1210.  
  1211. <------------->
  1212. --- (11 headers 0 lines) ---
  1213. Sending to 192.168.50.249 : 5060 (no NAT)
  1214.  
  1215. <--- Reliably Transmitting (no NAT) to 192.168.50.249:5060 --->
  1216. SIP/2.0 487 Request Terminated
  1217. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249
  1218. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  1219. To: <sip:2424@192.168.50.1;user=phone>;tag=as2c27759c
  1220. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  1221. CSeq: 2 INVITE
  1222. User-Agent: Asterisk PBX
  1223. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1224. Supported: replaces
  1225. Content-Length: 0
  1226.  
  1227.  
  1228. [Kpbx-b*CLI>
  1229. <------------>
  1230.  
  1231. <--- Transmitting (no NAT) to 192.168.50.249:5060 --->
  1232. SIP/2.0 200 OK
  1233. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249
  1234. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  1235. To: <sip:2424@192.168.50.1;user=phone>;tag=as2c27759c
  1236. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  1237. CSeq: 2 CANCEL
  1238. User-Agent: Asterisk PBX
  1239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  1240. Supported: replaces
  1241. Contact: <sip:2424@192.168.50.1>
  1242. Content-Length: 0
  1243.  
  1244.  
  1245. <------------>
  1246.  
  1247. [Kpbx-b*CLI>
  1248. [Sep 25 16:02:46] WARNING[11231]: channel.c:2557 ast_prod: Prodding channel 'SIP/2608-b7d05e10' failed
  1249.     -- Started music on hold, class 'ring', on SIP/2608-b7d05e10
  1250.  
  1251. [Kpbx-b*CLI>
  1252.     -- Executing [2604@standard:1] Set("Local/2604@standard-339e,2", "ARGS=dialExt,2604,no") in new stack
  1253.     -- Executing [2604@standard:2] AGI("Local/2604@standard-339e,2", "callRoute3.agi") in new stack
  1254.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  1255.     -- Executing [2605@standard:1] Set("Local/2605@standard-9a30,2", "ARGS=dialExt,2605,no") in new stack
  1256.     -- Executing [2605@standard:2] AGI("Local/2605@standard-9a30,2", "callRoute3.agi") in new stack
  1257.  
  1258. <--- SIP read from 192.168.50.249:5060 --->
  1259. ACK sip:2424@192.168.50.1 SIP/2.0
  1260. Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0
  1261. From: "Polycom 650" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
  1262. To: <sip:2424@192.168.50.1;user=phone>;tag=as2c27759c
  1263. CSeq: 2 ACK
  1264. Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
  1265. Contact: <sip:2608@192.168.50.249>
  1266. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  1267. User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084
  1268. Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="d940f8c753f5f514fc5d5a58dbbda37b", algorithm=MD5
  1269. Max-Forwards: 70
  1270. Content-Length: 0
  1271.  
  1272.  
  1273. <------------->
  1274. --- (12 headers 0 lines) ---
  1275.  
  1276. [Kpbx-b*CLI>
  1277.     -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi
  1278.     -- Stopped music on hold on SIP/2608-b7d05e10
  1279.  
  1280. [Kpbx-b*CLI>
  1281.   == Spawn extension (standard, 2604, 2) exited non-zero on 'Local/2604@standard-339e,2'
  1282.  
  1283. [Kpbx-b*CLI>
  1284.   == Spawn extension (standard, 2605, 2) exited non-zero on 'Local/2605@standard-9a30,2'
  1285.  
  1286. [Kpbx-b*CLI>
  1287.   == Spawn extension (standard, 2424, 1) exited non-zero on 'SIP/2608-b7d05e10'
  1288.  
  1289. [Kpbx-b*CLI>
  1290. Really destroying SIP dialog '6c7b529a-ba914749-fe3faecc@192.168.50.249' Method: ACK
  1291.  
  1292. [Kpbx-b*CLI>

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