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  1. sip.conf
  2. [itc-broadvoice-outbound]
  3. host=206.15.130.13
  4. fromdomain=206.15.130.13
  5. bindport=5060
  6. type=peer
  7. disallow=all
  8. allow=ulaw
  9. dtmfmode=rfc2833
  10. qualify=yes
  11. trustrpid=yes
  12. sendrpid=yes
  13.  
  14.  
  15. SIP DEBUG
  16.  
  17. itc-sip02-itc-com*CLI> sip set debug peer 3010
  18. SIP Debugging Enabled for IP: 192.168.7.14
  19. > 0x7faab400d100 -- Strict RTP learning complete - Locking on source address 98.152.0.243:50231
  20. [Jul 17 09:26:23] WARNING[10061]: chan_sip.c:4069 retrans_pkt: Retransmission timeout reached on transmission 1667690374-770352204-486373724 for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  21. Packet timed out after 31999ms with no response
  22. [Jul 17 09:26:23] WARNING[10061]: chan_sip.c:4128 retrans_pkt: Timeout on 1667690374-770352204-486373724 on non-critical invite transaction.
  23.  
  24. <--- SIP read from UDP:192.168.7.14:5060 --->
  25. INVITE sip:3013661953@itcsip02.itcurves.us:5060;user=phone SIP/2.0
  26. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK14515d3CE6CF8D4
  27. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  28. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>
  29. CSeq: 1 INVITE
  30. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  31. Contact: <sip:3010@192.168.7.14>
  32. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  33. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  34. Accept-Language: en
  35. Supported: 100rel,replaces
  36. Allow-Events: conference,talk,hold
  37. Max-Forwards: 70
  38. Content-Type: application/sdp
  39. Content-Length: 294
  40.  
  41. v=0
  42. o=- 1563369983 1563369983 IN IP4 192.168.7.14
  43. s=Polycom IP Phone
  44. c=IN IP4 192.168.7.14
  45. t=0 0
  46. a=sendrecv
  47. m=audio 2244 RTP/AVP 9 0 8 18 127
  48. a=rtpmap:9 G722/8000
  49. a=rtpmap:0 PCMU/8000
  50. a=rtpmap:8 PCMA/8000
  51. a=rtpmap:18 G729/8000
  52. a=fmtp:18 annexb=no
  53. a=rtpmap:127 telephone-event/8000
  54. <------------->
  55. --- (15 headers 13 lines) ---
  56. Sending to 192.168.7.14:5060 (no NAT)
  57. Sending to 192.168.7.14:5060 (no NAT)
  58. Using INVITE request as basis request - 49050705-b4fb6966-69b8f8f7@192.168.7.14
  59. Found peer '3010' for '3010' from 192.168.7.14:5060
  60.  
  61. <--- Reliably Transmitting (NAT) to 192.168.7.14:5060 --->
  62. SIP/2.0 401 Unauthorized
  63. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK14515d3CE6CF8D4;received=192.168.7.14;rport=5060
  64. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  65. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7a497eb2
  66. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  67. CSeq: 1 INVITE
  68. Server: Asterisk PBX 13.24.0
  69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  70. Supported: replaces, timer
  71. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5858962f"
  72. Content-Length: 0
  73.  
  74.  
  75. <------------>
  76. Scheduling destruction of SIP dialog '49050705-b4fb6966-69b8f8f7@192.168.7.14' in 6400 ms (Method: INVITE)
  77.  
  78. <--- SIP read from UDP:192.168.7.14:5060 --->
  79. ACK sip:3013661953@itcsip02.itcurves.us:5060;user=phone SIP/2.0
  80. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK14515d3CE6CF8D4
  81. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  82. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7a497eb2
  83. CSeq: 1 ACK
  84. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  85. Contact: <sip:3010@192.168.7.14>
  86. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  87. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  88. Accept-Language: en
  89. Max-Forwards: 70
  90. Content-Length: 0
  91.  
  92. <------------->
  93. --- (12 headers 0 lines) ---
  94.  
  95. <--- SIP read from UDP:192.168.7.14:5060 --->
  96. INVITE sip:3013661953@itcsip02.itcurves.us:5060;user=phone SIP/2.0
  97. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK81b23eb8402873A9
  98. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  99. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>
  100. CSeq: 2 INVITE
  101. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  102. Contact: <sip:3010@192.168.7.14>
  103. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  104. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  105. Accept-Language: en
  106. Supported: 100rel,replaces
  107. Allow-Events: conference,talk,hold
  108. Authorization: Digest username="3010", realm="asterisk", nonce="5858962f", uri="sip:3013661953@itcsip02.itcurves.us:5060;user=phone", response="517b6b98cfa513fcc45b8407a3c66bf4", algorithm=MD5
  109. Max-Forwards: 70
  110. Content-Type: application/sdp
  111. Content-Length: 294
  112.  
  113. v=0
  114. o=- 1563369983 1563369983 IN IP4 192.168.7.14
  115. s=Polycom IP Phone
  116. c=IN IP4 192.168.7.14
  117. t=0 0
  118. a=sendrecv
  119. m=audio 2244 RTP/AVP 9 0 8 18 127
  120. a=rtpmap:9 G722/8000
  121. a=rtpmap:0 PCMU/8000
  122. a=rtpmap:8 PCMA/8000
  123. a=rtpmap:18 G729/8000
  124. a=fmtp:18 annexb=no
  125. a=rtpmap:127 telephone-event/8000
  126. <------------->
  127. --- (16 headers 13 lines) ---
  128. Sending to 192.168.7.14:5060 (NAT)
  129. Using INVITE request as basis request - 49050705-b4fb6966-69b8f8f7@192.168.7.14
  130. Found peer '3010' for '3010' from 192.168.7.14:5060
  131. == Using SIP RTP CoS mark 5
  132. Found RTP audio format 9
  133. Found RTP audio format 0
  134. Found RTP audio format 8
  135. Found RTP audio format 18
  136. Found RTP audio format 127
  137. Found audio description format G722 for ID 9
  138. Found audio description format PCMU for ID 0
  139. Found audio description format PCMA for ID 8
  140. Found audio description format G729 for ID 18
  141. Found audio description format telephone-event for ID 127
  142. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  143. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  144. > 0x7fa9d830c120 -- Strict RTP learning after remote address set to: 192.168.7.14:2244
  145. Peer audio RTP is at port 192.168.7.14:2244
  146. Looking for 3013661953 in taxi-us-agent/n (domain itcsip02.itcurves.us)
  147. sip_route_dump: route/path hop: <sip:3010@192.168.7.14>
  148.  
  149. <--- Transmitting (NAT) to 192.168.7.14:5060 --->
  150. SIP/2.0 100 Trying
  151. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK81b23eb8402873A9;received=192.168.7.14;rport=5060
  152. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  153. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>
  154. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  155. CSeq: 2 INVITE
  156. Server: Asterisk PBX 13.24.0
  157. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  158. Supported: replaces, timer
  159. Contact: <sip:3013661953@192.168.15.12:5060>
  160. Content-Length: 0
  161.  
  162.  
  163. <------------>
  164. -- Executing [3013661953@taxi-us-agent/n:1] Set("SIP/3010-00009e00", "CDR(userfield)=Agent-Outgoing") in new stack
  165. -- Executing [3013661953@taxi-us-agent/n:2] Set("SIP/3010-00009e00", "CALLERID(num)=8886754545") in new stack
  166. -- Executing [3013661953@taxi-us-agent/n:3] Dial("SIP/3010-00009e00", "SIP/3013661953@itc-broadvoice-outbound,,t") in new stack
  167. == Using SIP RTP CoS mark 5
  168. -- Called SIP/3013661953@itc-broadvoice-outbound
  169.  
  170. Reliably Transmitting (NAT) to 192.168.7.14:5060:
  171. OPTIONS sip:3010@192.168.7.14 SIP/2.0
  172. Via: SIP/2.0/UDP 192.168.15.12:5060;branch=z9hG4bK39e1654d;rport
  173. Max-Forwards: 70
  174. From: "asterisk" <sip:asterisk@192.168.15.12>;tag=as3eba39ff
  175. To: <sip:3010@192.168.7.14>
  176. Contact: <sip:asterisk@192.168.15.12:5060>
  177. Call-ID: 0872c9ae2212a67f6596d3990ad1a941@192.168.15.12:5060
  178. CSeq: 102 OPTIONS
  179. User-Agent: Asterisk PBX 13.24.0
  180. Date: Wed, 17 Jul 2019 13:26:25 GMT
  181. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  182. Supported: replaces, timer
  183. Content-Length: 0
  184.  
  185.  
  186. ---
  187.  
  188. <--- SIP read from UDP:192.168.7.14:5060 --->
  189. SIP/2.0 200 OK
  190. Via: SIP/2.0/UDP 192.168.15.12:5060;branch=z9hG4bK39e1654d;rport
  191. From: "asterisk" <sip:asterisk@192.168.15.12>;tag=as3eba39ff
  192. To: "3010" <sip:3010@192.168.7.14>;tag=41EF234D-860A6B2E
  193. CSeq: 102 OPTIONS
  194. Call-ID: 0872c9ae2212a67f6596d3990ad1a941@192.168.15.12:5060
  195. Contact: <sip:3010@192.168.7.14>
  196. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  197. Supported: 100rel,replaces,100rel,timer,replaces,norefersub
  198. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  199. Accept-Language: en
  200. Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
  201. Accept-Encoding: identity
  202. Content-Length: 0
  203.  
  204. <------------->
  205. --- (14 headers 0 lines) ---
  206. Really destroying SIP dialog '0872c9ae2212a67f6596d3990ad1a941@192.168.15.12:5060' Method: OPTIONS
  207. > 0x7fa998007510 -- Strict RTP learning after remote address set to: 206.15.130.13:46556
  208. -- SIP/itc-broadvoice-outbound-00009e01 is making progress passing it to SIP/3010-00009e00
  209. Audio is at 35514
  210. Adding codec ulaw to SDP
  211. Adding codec alaw to SDP
  212. Adding non-codec 0x1 (telephone-event) to SDP
  213.  
  214. <--- Transmitting (NAT) to 192.168.7.14:5060 --->
  215. SIP/2.0 183 Session Progress
  216. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK81b23eb8402873A9;received=192.168.7.14;rport=5060
  217. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  218. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7d6f7747
  219. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  220. CSeq: 2 INVITE
  221. Server: Asterisk PBX 13.24.0
  222. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  223. Supported: replaces, timer
  224. Contact: <sip:3013661953@192.168.15.12:5060>
  225. Content-Type: application/sdp
  226. Content-Length: 266
  227.  
  228. v=0
  229. o=root 1600988103 1600988103 IN IP4 192.168.15.12
  230. s=Asterisk PBX 13.24.0
  231. c=IN IP4 192.168.15.12
  232. t=0 0
  233. m=audio 35514 RTP/AVP 0 8 127
  234. a=rtpmap:0 PCMU/8000
  235. a=rtpmap:8 PCMA/8000
  236. a=rtpmap:127 telephone-event/8000
  237. a=fmtp:127 0-16
  238. a=maxptime:150
  239. a=sendrecv
  240.  
  241. <------------>
  242. > 0x7fa998007510 -- Strict RTP switching to RTP target address 206.15.130.13:46556 as source
  243. > 0x7fa9d830c120 -- Strict RTP switching to RTP target address 192.168.7.14:2244 as source
  244. > 0x7fa9d830c120 -- Strict RTP learning complete - Locking on source address 192.168.7.14:2244
  245. -- User entered nothing, 1 chance left
  246. -- <SIP/itc-vitel-inbound-00009dfe> Playing '/scripts/taxi_us/recordings/2132845440-1563369967.326665new.slin' (language 'en')
  247. -- SIP/itc-broadvoice-outbound-00009e01 answered SIP/3010-00009e00
  248. Audio is at 35514
  249. Adding codec ulaw to SDP
  250. Adding codec alaw to SDP
  251. Adding non-codec 0x1 (telephone-event) to SDP
  252.  
  253. <--- Reliably Transmitting (NAT) to 192.168.7.14:5060 --->
  254. SIP/2.0 200 OK
  255. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK81b23eb8402873A9;received=192.168.7.14;rport=5060
  256. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  257. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7d6f7747
  258. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  259. CSeq: 2 INVITE
  260. Server: Asterisk PBX 13.24.0
  261. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  262. Supported: replaces, timer
  263. Contact: <sip:3013661953@192.168.15.12:5060>
  264. Content-Type: application/sdp
  265. Content-Length: 266
  266.  
  267. v=0
  268. o=root 1600988103 1600988103 IN IP4 192.168.15.12
  269. s=Asterisk PBX 13.24.0
  270. c=IN IP4 192.168.15.12
  271. t=0 0
  272. m=audio 35514 RTP/AVP 0 8 127
  273. a=rtpmap:0 PCMU/8000
  274. a=rtpmap:8 PCMA/8000
  275. a=rtpmap:127 telephone-event/8000
  276. a=fmtp:127 0-16
  277. a=maxptime:150
  278. a=sendrecv
  279.  
  280. <------------>
  281. -- Channel SIP/itc-broadvoice-outbound-00009e01 joined 'simple_bridge' basic-bridge <d494d2ea-2c98-4a4c-8e0f-449c928f233f>
  282. -- Channel SIP/3010-00009e00 joined 'simple_bridge' basic-bridge <d494d2ea-2c98-4a4c-8e0f-449c928f233f>
  283.  
  284. <--- SIP read from UDP:192.168.7.14:5060 --->
  285. ACK sip:3013661953@192.168.15.12:5060 SIP/2.0
  286. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bKc6cc683f4FEB2380
  287. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  288. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7d6f7747
  289. CSeq: 2 ACK
  290. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  291. Contact: <sip:3010@192.168.7.14>
  292. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  293. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  294. Accept-Language: en
  295. Max-Forwards: 70
  296. Content-Length: 0
  297.  
  298. <------------->
  299. --- (12 headers 0 lines) ---
  300. > 0x7fa998007510 -- Strict RTP learning complete - Locking on source address 206.15.130.13:46556
  301.  
  302. <--- SIP read from UDP:192.168.7.14:5060 --->
  303. BYE sip:3013661953@192.168.15.12:5060 SIP/2.0
  304. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK7ab155f16F6892
  305. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  306. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7d6f7747
  307. CSeq: 3 BYE
  308. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  309. Contact: <sip:3010@192.168.7.14>
  310. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  311. Accept-Language: en
  312. Authorization: Digest username="3010", realm="asterisk", nonce="5858962f", uri="sip:3013661953@itcsip02.itcurves.us:5060;user=phone", response="d096a6559bb61e737ee79fb68fc786ab", algorithm=MD5
  313. Max-Forwards: 70
  314. Content-Length: 0
  315.  
  316. <------------->
  317. --- (12 headers 0 lines) ---
  318. Sending to 192.168.7.14:5060 (NAT)
  319. Scheduling destruction of SIP dialog '49050705-b4fb6966-69b8f8f7@192.168.7.14' in 6400 ms (Method: BYE)
  320.  
  321. <--- Transmitting (NAT) to 192.168.7.14:5060 --->
  322. SIP/2.0 200 OK
  323. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK7ab155f16F6892;received=192.168.7.14;rport=5060
  324. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=6DC9D61-2EB8E502
  325. To: <sip:3013661953@itcsip02.itcurves.us;user=phone>;tag=as7d6f7747
  326. Call-ID: 49050705-b4fb6966-69b8f8f7@192.168.7.14
  327. CSeq: 3 BYE
  328. Server: Asterisk PBX 13.24.0
  329. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  330. Supported: replaces, timer
  331. Content-Length: 0
  332.  
  333.  
  334. <------------>
  335. -- Channel SIP/3010-00009e00 left 'simple_bridge' basic-bridge <d494d2ea-2c98-4a4c-8e0f-449c928f233f>
  336. -- Channel SIP/itc-broadvoice-outbound-00009e01 left 'simple_bridge' basic-bridge <d494d2ea-2c98-4a4c-8e0f-449c928f233f>
  337. == Spawn extension (taxi-us-agent/n, 3013661953, 6) exited non-zero on 'SIP/3010-00009e00'
  338. -- Channel SIP/3004-00009dff left 'simple_bridge' basic-bridge <13797edb-2978-4f5d-9f6f-9e6ad4d16e15>
  339. -- Channel SIP/itc-vitel-inbound-00009dfd left 'simple_bridge' basic-bridge <13797edb-2978-4f5d-9f6f-9e6ad4d16e15>
  340. == Spawn extension (502-csa-korean, 0, 5) exited non-zero on 'SIP/itc-vitel-inbound-00009dfd'
  341.  
  342. <--- SIP read from UDP:192.168.7.14:5060 --->
  343. REGISTER sip:itcsip02.itcurves.us:5060 SIP/2.0
  344. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK2d1c7463B274264
  345. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=4DF98B13-F4F52A14
  346. To: <sip:3010@itcsip02.itcurves.us>
  347. CSeq: 4645 REGISTER
  348. Call-ID: da97de37-629dff8-eeb010e9@192.168.7.14
  349. Contact: <sip:3010@192.168.7.14>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
  350. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  351. Accept-Language: en
  352. Authorization: Digest username="3010", realm="asterisk", nonce="6902d74e", uri="sip:itcsip02.itcurves.us:5060", response="d972604636176014c610afa99b5ef9c0", algorithm=MD5
  353. Max-Forwards: 70
  354. Expires: 30
  355. Content-Length: 0
  356.  
  357. <------------->
  358. --- (13 headers 0 lines) ---
  359. Sending to 192.168.7.14:5060 (no NAT)
  360. [Jul 17 09:26:36] NOTICE[10061]: chan_sip.c:17310 check_auth: Correct auth, but based on stale nonce received from '"3010" <sip:3010@itcsip02.itcurves.us>;tag=4DF98B13-F4F52A14'
  361.  
  362. <--- Transmitting (NAT) to 192.168.7.14:5060 --->
  363. SIP/2.0 401 Unauthorized
  364. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK2d1c7463B274264;received=192.168.7.14;rport=5060
  365. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=4DF98B13-F4F52A14
  366. To: <sip:3010@itcsip02.itcurves.us>;tag=as3295fe8d
  367. Call-ID: da97de37-629dff8-eeb010e9@192.168.7.14
  368. CSeq: 4645 REGISTER
  369. Server: Asterisk PBX 13.24.0
  370. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  371. Supported: replaces, timer
  372. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f6fe007", stale=true
  373. Content-Length: 0
  374.  
  375.  
  376. <------------>
  377. Scheduling destruction of SIP dialog 'da97de37-629dff8-eeb010e9@192.168.7.14' in 32000 ms (Method: REGISTER)
  378.  
  379. <--- SIP read from UDP:192.168.7.14:5060 --->
  380. REGISTER sip:itcsip02.itcurves.us:5060 SIP/2.0
  381. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK51774b958C9DB8F6
  382. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=4DF98B13-F4F52A14
  383. To: <sip:3010@itcsip02.itcurves.us>
  384. CSeq: 4646 REGISTER
  385. Call-ID: da97de37-629dff8-eeb010e9@192.168.7.14
  386. Contact: <sip:3010@192.168.7.14>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
  387. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  388. Accept-Language: en
  389. Authorization: Digest username="3010", realm="asterisk", nonce="4f6fe007", uri="sip:itcsip02.itcurves.us:5060", response="eff13fdae149e54def08967a426cf71e", algorithm=MD5
  390. Max-Forwards: 70
  391. Expires: 30
  392. Content-Length: 0
  393.  
  394. <------------->
  395. --- (13 headers 0 lines) ---
  396. Sending to 192.168.7.14:5060 (no NAT)
  397. Reliably Transmitting (NAT) to 192.168.7.14:5060:
  398. OPTIONS sip:3010@192.168.7.14 SIP/2.0
  399. Via: SIP/2.0/UDP 192.168.15.12:5060;branch=z9hG4bK2485b58a;rport
  400. Max-Forwards: 70
  401. From: "asterisk" <sip:asterisk@192.168.15.12>;tag=as440a5397
  402. To: <sip:3010@192.168.7.14>
  403. Contact: <sip:asterisk@192.168.15.12:5060>
  404. Call-ID: 3335dcf072e40859091884c810e39633@192.168.15.12:5060
  405. CSeq: 102 OPTIONS
  406. User-Agent: Asterisk PBX 13.24.0
  407. Date: Wed, 17 Jul 2019 13:26:36 GMT
  408. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  409. Supported: replaces, timer
  410. Content-Length: 0
  411.  
  412.  
  413. ---
  414.  
  415. <--- Transmitting (NAT) to 192.168.7.14:5060 --->
  416. SIP/2.0 200 OK
  417. Via: SIP/2.0/UDP 192.168.7.14;branch=z9hG4bK51774b958C9DB8F6;received=192.168.7.14;rport=5060
  418. From: "3010" <sip:3010@itcsip02.itcurves.us>;tag=4DF98B13-F4F52A14
  419. To: <sip:3010@itcsip02.itcurves.us>;tag=as3295fe8d
  420. Call-ID: da97de37-629dff8-eeb010e9@192.168.7.14
  421. CSeq: 4646 REGISTER
  422. Server: Asterisk PBX 13.24.0
  423. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  424. Supported: replaces, timer
  425. Expires: 60
  426. Contact: <sip:3010@192.168.7.14>;expires=60
  427. Date: Wed, 17 Jul 2019 13:26:36 GMT
  428. Content-Length: 0
  429.  
  430.  
  431. <------------>
  432. Scheduling destruction of SIP dialog 'da97de37-629dff8-eeb010e9@192.168.7.14' in 32000 ms (Method: REGISTER)
  433.  
  434. <--- SIP read from UDP:192.168.7.14:5060 --->
  435. SIP/2.0 200 OK
  436. Via: SIP/2.0/UDP 192.168.15.12:5060;branch=z9hG4bK2485b58a;rport
  437. From: "asterisk" <sip:asterisk@192.168.15.12>;tag=as440a5397
  438. To: "3010" <sip:3010@192.168.7.14>;tag=CD78D448-CBF94439
  439. CSeq: 102 OPTIONS
  440. Call-ID: 3335dcf072e40859091884c810e39633@192.168.15.12:5060
  441. Contact: <sip:3010@192.168.7.14>
  442. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  443. Supported: 100rel,replaces,100rel,timer,replaces,norefersub
  444. User-Agent: PolycomSoundPointIP-SPIP_670-UA/4.0.14.0987
  445. Accept-Language: en
  446. Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
  447. Accept-Encoding: identity
  448. Content-Length: 0
  449.  
  450. <------------->
  451. --- (14 headers 0 lines) ---
  452. Really destroying SIP dialog '3335dcf072e40859091884c810e39633@192.168.15.12:5060' Method: OPTIONS
  453. itc-sip02-itc-com*CLI> sip set debug off
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