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jren207
Oct 26th, 2017
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  1. jren207@asterisk:/etc/asterisk$ sudo asterisk -rvvvvv
  2. Asterisk 15.0.0, Copyright (C) 1999 - 2016, Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 15.0.0 currently running on asterisk (pid = 1243)
  10. asterisk*CLI> pjsip set logger on
  11. PJSIP Logging enabled
  12. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  13. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  14. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  15. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  16. To: <sip:1002@10.0.0.12>
  17. Contact: <sip:1002@10.0.0.16:5060>
  18. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  19. CSeq: 53198 OPTIONS
  20. Max-Forwards: 70
  21. User-Agent: Asterisk PBX 15.0.0
  22. Content-Length: 0
  23.  
  24.  
  25. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  26. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  27. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  28. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  29. To: <sip:1002@10.0.0.12>
  30. Contact: <sip:1002@10.0.0.16:5060>
  31. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  32. CSeq: 53198 OPTIONS
  33. Max-Forwards: 70
  34. User-Agent: Asterisk PBX 15.0.0
  35. Content-Length: 0
  36.  
  37.  
  38. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  39. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  40. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  41. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  42. To: <sip:1002@10.0.0.12>
  43. Contact: <sip:1002@10.0.0.16:5060>
  44. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  45. CSeq: 53198 OPTIONS
  46. Max-Forwards: 70
  47. User-Agent: Asterisk PBX 15.0.0
  48. Content-Length: 0
  49.  
  50.  
  51. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  52. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  53. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPja23383a7-3793-46e5-b4cb-d0e4379fceb3
  54. From: <sip:1002@10.0.0.16>;tag=a575bc83-b055-4443-a013-fb0df0b02197
  55. To: <sip:1002@10.0.0.12>
  56. Contact: <sip:1002@10.0.0.16:5060>
  57. Call-ID: 46b11e71-3154-48ac-b3ca-67da50072ef6
  58. CSeq: 61379 OPTIONS
  59. Max-Forwards: 70
  60. User-Agent: Asterisk PBX 15.0.0
  61. Content-Length: 0
  62.  
  63.  
  64. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  65. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  66. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  67. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  68. To: <sip:1002@10.0.0.12>
  69. Contact: <sip:1002@10.0.0.16:5060>
  70. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  71. CSeq: 53198 OPTIONS
  72. Max-Forwards: 70
  73. User-Agent: Asterisk PBX 15.0.0
  74. Content-Length: 0
  75.  
  76.  
  77. <--- Received SIP request (1390 bytes) from UDP:217.10.68.151:5060 --->
  78. INVITE sip:NUMBERTRUNK@10.0.0.16:5060 SIP/2.0
  79. Record-Route: <sip:217.10.68.151;lr;ftag=as1bc1d144>
  80. Record-Route: <sip:172.20.40.5;lr>
  81. Record-Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  82. Via: SIP/2.0/UDP 217.10.68.151;branch=z9hG4bK47d7.be2dbec7fdc434f8be91ef69fba91753.0
  83. Via: SIP/2.0/UDP 172.20.40.5;branch=z9hG4bK47d7.79d4d3f4c150474efd3c9304fac2a16b.0
  84. Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK47d7.0b6198fa060a2ff2fccc32f9d915c079.0
  85. Via: SIP/2.0/UDP 217.116.117.7:5060;branch=z9hG4bK07ca50af
  86. Max-Forwards: 67
  87. From: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  88. To: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>
  89. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  90. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  91. CSeq: 103 INVITE
  92. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  93. Supported: replaces
  94. Content-Type: application/sdp
  95. Content-Length: 443
  96.  
  97. v=0
  98. o=root 1607684837 1607684838 IN IP4 217.116.117.7
  99. s=sipgate VoIP GW
  100. c=IN IP4 212.9.44.251
  101. t=0 0
  102. m=audio 22750 RTP/AVP 8 0 3 97 18 112 101
  103. a=rtpmap:8 PCMA/8000
  104. a=rtpmap:0 PCMU/8000
  105. a=rtpmap:3 GSM/8000
  106. a=rtpmap:97 iLBC/8000
  107. a=fmtp:97 mode=30
  108. a=rtpmap:18 G729/8000
  109. a=fmtp:18 annexb=no
  110. a=rtpmap:112 G726-32/8000
  111. a=rtpmap:101 telephone-event/8000
  112. a=fmtp:101 0-16
  113. a=silenceSupp:off - - - -
  114. a=ptime:20
  115. a=sendrecv
  116. a=rtcp:22751
  117.  
  118. == Setting global variable 'SIPDOMAIN' to '10.0.0.16'
  119. <--- Transmitting SIP response (786 bytes) to UDP:217.10.68.151:5060 --->
  120. SIP/2.0 100 Trying
  121. Via: SIP/2.0/UDP 217.10.68.151;rport=5060;received=217.10.68.151;branch=z9hG4bK47d7.be2dbec7fdc434f8be91ef69fba91753.0
  122. Via: SIP/2.0/UDP 172.20.40.5;branch=z9hG4bK47d7.79d4d3f4c150474efd3c9304fac2a16b.0
  123. Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK47d7.0b6198fa060a2ff2fccc32f9d915c079.0
  124. Via: SIP/2.0/UDP 217.116.117.7:5060;branch=z9hG4bK07ca50af
  125. Record-Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  126. Record-Route: <sip:172.20.40.5;lr>
  127. Record-Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  128. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  129. From: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  130. To: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>
  131. CSeq: 103 INVITE
  132. Server: Asterisk PBX 15.0.0
  133. Content-Length: 0
  134.  
  135.  
  136. -- Executing [NUMBERTRUNK@from-external:1] NoOp("PJSIP/sipgate-00000006", "'Sipgate External Call'") in new stack
  137. -- Executing [NUMBERTRUNK@from-external:2] Set("PJSIP/sipgate-00000006", "CALLERID(name)=External") in new stack
  138. -- Executing [NUMBERTRUNK@from-external:3] Dial("PJSIP/sipgate-00000006", "PJSIP/1001") in new stack
  139. -- Called PJSIP/1001
  140. <--- Transmitting SIP request (1072 bytes) to UDP:10.0.0.30:5060 --->
  141. INVITE sip:1001@10.0.0.30:5060 SIP/2.0
  142. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj08b6d801-8e8f-4623-8f1a-1287f92b039e
  143. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  144. To: <sip:1001@10.0.0.30>
  145. Contact: <sip:asterisk@10.0.0.16:5060>
  146. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  147. CSeq: 14997 INVITE
  148. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  149. Supported: 100rel, timer, replaces, norefersub
  150. Session-Expires: 1800
  151. Min-SE: 90
  152. P-Asserted-Identity: "External" <sip:NUMBERMOBILE@10.0.0.16>
  153. Remote-Party-ID: "External" <sip:NUMBERMOBILE@10.0.0.16>;privacy=off;screen=no
  154. Max-Forwards: 70
  155. User-Agent: Asterisk PBX 15.0.0
  156. Content-Type: application/sdp
  157. Content-Length: 279
  158.  
  159. v=0
  160. o=- 1393429158 1393429158 IN IP4 10.0.0.16
  161. s=Asterisk
  162. c=IN IP4 10.0.0.16
  163. t=0 0
  164. m=audio 11542 RTP/AVP 0 8 9 101
  165. a=rtpmap:0 PCMU/8000
  166. a=rtpmap:8 PCMA/8000
  167. a=rtpmap:9 G722/8000
  168. a=rtpmap:101 telephone-event/8000
  169. a=fmtp:101 0-16
  170. a=ptime:20
  171. a=maxptime:150
  172. a=sendrecv
  173.  
  174. <--- Received SIP response (389 bytes) from UDP:10.0.0.30:5060 --->
  175. SIP/2.0 100 Trying
  176. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj08b6d801-8e8f-4623-8f1a-1287f92b039e;received=10.0.0.16
  177. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  178. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  179. To: <sip:1001@10.0.0.30>
  180. CSeq: 14997 INVITE
  181. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  182. Content-Length: 0
  183.  
  184.  
  185. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  186. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  187. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  188. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  189. To: <sip:1002@10.0.0.12>
  190. Contact: <sip:1002@10.0.0.16:5060>
  191. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  192. CSeq: 53198 OPTIONS
  193. Max-Forwards: 70
  194. User-Agent: Asterisk PBX 15.0.0
  195. Content-Length: 0
  196.  
  197.  
  198. <--- Received SIP response (553 bytes) from UDP:10.0.0.30:5060 --->
  199. SIP/2.0 180 Ringing
  200. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj08b6d801-8e8f-4623-8f1a-1287f92b039e;received=10.0.0.16
  201. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  202. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  203. To: <sip:1001@10.0.0.30>;tag=1252544688
  204. CSeq: 14997 INVITE
  205. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  206. Contact: <sip:1001@10.0.0.30:5060>
  207. Require: 100rel
  208. RSeq: 1206104165
  209. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  210. Content-Length: 0
  211.  
  212.  
  213. <--- Transmitting SIP request (429 bytes) to UDP:10.0.0.30:5060 --->
  214. PRACK sip:1001@10.0.0.30:5060 SIP/2.0
  215. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj8778fb23-a016-4d8b-a71b-bbb4fb27a40a
  216. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  217. To: <sip:1001@10.0.0.30>;tag=1252544688
  218. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  219. CSeq: 14998 PRACK
  220. RAck: 1206104165 14997 INVITE
  221. Max-Forwards: 70
  222. User-Agent: Asterisk PBX 15.0.0
  223. Content-Length: 0
  224.  
  225.  
  226. -- PJSIP/1001-00000007 is ringing
  227. <--- Transmitting SIP response (970 bytes) to UDP:217.10.68.151:5060 --->
  228. SIP/2.0 180 Ringing
  229. Via: SIP/2.0/UDP 217.10.68.151;rport=5060;received=217.10.68.151;branch=z9hG4bK47d7.be2dbec7fdc434f8be91ef69fba91753.0
  230. Via: SIP/2.0/UDP 172.20.40.5;branch=z9hG4bK47d7.79d4d3f4c150474efd3c9304fac2a16b.0
  231. Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK47d7.0b6198fa060a2ff2fccc32f9d915c079.0
  232. Via: SIP/2.0/UDP 217.116.117.7:5060;branch=z9hG4bK07ca50af
  233. Record-Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  234. Record-Route: <sip:172.20.40.5;lr>
  235. Record-Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  236. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  237. From: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  238. To: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  239. CSeq: 103 INVITE
  240. Server: Asterisk PBX 15.0.0
  241. Contact: <sip:10.0.0.16:5060>
  242. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  243. Content-Length: 0
  244.  
  245.  
  246. <--- Received SIP response (476 bytes) from UDP:10.0.0.30:5060 --->
  247. SIP/2.0 200 OK
  248. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj8778fb23-a016-4d8b-a71b-bbb4fb27a40a;received=10.0.0.16
  249. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  250. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  251. To: <sip:1001@10.0.0.30>;tag=1252544688
  252. CSeq: 14998 PRACK
  253. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  254. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  255. Content-Length: 0
  256.  
  257.  
  258. <--- Received SIP response (805 bytes) from UDP:10.0.0.30:5060 --->
  259. SIP/2.0 200 OK
  260. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj08b6d801-8e8f-4623-8f1a-1287f92b039e;received=10.0.0.16
  261. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  262. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  263. To: <sip:1001@10.0.0.30>;tag=1252544688
  264. CSeq: 14997 INVITE
  265. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  266. Supported: 100rel,precondition
  267. Contact: <sip:1001@10.0.0.30:5060>
  268. Require: replaces
  269. Content-Type: application/sdp
  270. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  271. Content-Length: 208
  272.  
  273. v=0
  274. o=- 1509055335 1509055335 IN IP4 10.0.0.30
  275. s=-
  276. c=IN IP4 10.0.0.30
  277. t=0 0
  278. m=audio 16022 RTP/AVP 0 101
  279. a=rtpmap:0 PCMU/8000
  280. a=rtpmap:101 telephone-event/8000
  281. a=fmtp:101 0-15
  282. a=sendrecv
  283. a=ptime:20
  284.  
  285. > 0x7f3444043da0 -- Strict RTP learning after remote address set to: 10.0.0.30:16022
  286. <--- Transmitting SIP request (394 bytes) to UDP:10.0.0.30:5060 --->
  287. ACK sip:1001@10.0.0.30:5060 SIP/2.0
  288. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj8e3a5ede-c367-42a0-b4e9-e72838fed360
  289. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  290. To: <sip:1001@10.0.0.30>;tag=1252544688
  291. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  292. CSeq: 14997 ACK
  293. Max-Forwards: 70
  294. User-Agent: Asterisk PBX 15.0.0
  295. Content-Length: 0
  296.  
  297.  
  298. -- PJSIP/1001-00000007 answered PJSIP/sipgate-00000006
  299. > 0x7f34440272b0 -- Strict RTP learning after remote address set to: 212.9.44.251:22750
  300. <--- Transmitting SIP response (1302 bytes) to UDP:217.10.68.151:5060 --->
  301. SIP/2.0 200 OK
  302. Via: SIP/2.0/UDP 217.10.68.151;rport=5060;received=217.10.68.151;branch=z9hG4bK47d7.be2dbec7fdc434f8be91ef69fba91753.0
  303. Via: SIP/2.0/UDP 172.20.40.5;branch=z9hG4bK47d7.79d4d3f4c150474efd3c9304fac2a16b.0
  304. Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK47d7.0b6198fa060a2ff2fccc32f9d915c079.0
  305. Via: SIP/2.0/UDP 217.116.117.7:5060;branch=z9hG4bK07ca50af
  306. Record-Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  307. Record-Route: <sip:172.20.40.5;lr>
  308. Record-Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  309. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  310. From: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  311. To: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  312. CSeq: 103 INVITE
  313. Server: Asterisk PBX 15.0.0
  314. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  315. Contact: <sip:10.0.0.16:5060>
  316. Supported: 100rel, timer, replaces, norefersub
  317. Content-Type: application/sdp
  318. Content-Length: 255
  319.  
  320. v=0
  321. o=- 1607684837 1607684840 IN IP4 10.0.0.16
  322. s=Asterisk
  323. c=IN IP4 10.0.0.16
  324. t=0 0
  325. m=audio 17714 RTP/AVP 0 8 101
  326. a=rtpmap:0 PCMU/8000
  327. a=rtpmap:8 PCMA/8000
  328. a=rtpmap:101 telephone-event/8000
  329. a=fmtp:101 0-16
  330. a=ptime:20
  331. a=maxptime:150
  332. a=sendrecv
  333.  
  334. -- Channel PJSIP/1001-00000007 joined 'simple_bridge' basic-bridge <4b5c2154-0be9-4d56-a58b-3bc2e3b99d8a>
  335. -- Channel PJSIP/sipgate-00000006 joined 'simple_bridge' basic-bridge <4b5c2154-0be9-4d56-a58b-3bc2e3b99d8a>
  336. > Bridge 4b5c2154-0be9-4d56-a58b-3bc2e3b99d8a: switching from simple_bridge technology to native_rtp
  337. > Remotely bridged 'PJSIP/sipgate-00000006' and 'PJSIP/1001-00000007' - media will flow directly between them
  338. <--- Transmitting SIP request (1042 bytes) to UDP:10.0.0.30:5060 --->
  339. INVITE sip:1001@10.0.0.30:5060 SIP/2.0
  340. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjc2ba0de0-9fb2-4027-9773-56c6e453a5bc
  341. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  342. To: <sip:1001@10.0.0.30>;tag=1252544688
  343. Contact: <sip:asterisk@10.0.0.16:5060>
  344. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  345. CSeq: 14999 INVITE
  346. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  347. Supported: 100rel, timer, replaces, norefersub
  348. Session-Expires: 1800
  349. Min-SE: 90
  350. P-Asserted-Identity: "External" <sip:NUMBERMOBILE@10.0.0.16>
  351. Remote-Party-ID: "External" <sip:NUMBERMOBILE@10.0.0.16>;privacy=off;screen=no
  352. Max-Forwards: 70
  353. User-Agent: Asterisk PBX 15.0.0
  354. Content-Type: application/sdp
  355. Content-Length: 234
  356.  
  357. v=0
  358. o=- 1393429158 1393429159 IN IP4 10.0.0.16
  359. s=Asterisk
  360. c=IN IP4 212.9.44.251
  361. t=0 0
  362. m=audio 22750 RTP/AVP 0 101
  363. a=rtpmap:0 PCMU/8000
  364. a=rtpmap:101 telephone-event/8000
  365. a=fmtp:101 0-16
  366. a=ptime:20
  367. a=maxptime:150
  368. a=sendrecv
  369.  
  370. <--- Received SIP request (709 bytes) from UDP:217.10.68.151:5060 --->
  371. ACK sip:10.0.0.16:5060 SIP/2.0
  372. Via: SIP/2.0/UDP 217.10.68.151;branch=z9hG4bK47d7.1b40a1876506783d0be77cdf4ae7bcd0.0
  373. Via: SIP/2.0/UDP 172.20.40.5;branch=z9hG4bK47d7.f8f1ba0f0d03eefab49ac52b2dad28e5.0
  374. Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK47d7.6345ad94d03688536c487a79820bd08f.0
  375. Via: SIP/2.0/UDP 217.116.117.7:5060;branch=z9hG4bK1b4988b0
  376. Max-Forwards: 67
  377. From: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  378. To: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  379. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  380. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  381. CSeq: 103 ACK
  382. Content-Length: 0
  383. X-hint: rr-enforced
  384.  
  385.  
  386. <--- Transmitting SIP request (1093 bytes) to UDP:217.10.68.151:5060 --->
  387. INVITE sip:NUMBERMOBILE@217.116.117.7:5060 SIP/2.0
  388. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  389. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  390. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  391. Contact: <sip:10.0.0.16:5060>
  392. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  393. CSeq: 19057 INVITE
  394. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  395. Route: <sip:172.20.40.5;lr>
  396. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  397. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  398. Supported: 100rel, timer, replaces, norefersub
  399. Session-Expires: 1800
  400. Min-SE: 90
  401. Max-Forwards: 70
  402. User-Agent: Asterisk PBX 15.0.0
  403. Content-Type: application/sdp
  404. Content-Length: 231
  405.  
  406. v=0
  407. o=- 1607684837 1607684841 IN IP4 10.0.0.16
  408. s=Asterisk
  409. c=IN IP4 10.0.0.30
  410. t=0 0
  411. m=audio 16022 RTP/AVP 0 101
  412. a=rtpmap:0 PCMU/8000
  413. a=rtpmap:101 telephone-event/8000
  414. a=fmtp:101 0-16
  415. a=ptime:20
  416. a=maxptime:150
  417. a=sendrecv
  418.  
  419. <--- Received SIP response (452 bytes) from UDP:217.10.68.151:5060 --->
  420. SIP/2.0 100 trying -- your call is important to us
  421. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  422. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  423. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  424. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  425. CSeq: 19057 INVITE
  426. Content-Length: 0
  427.  
  428.  
  429. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  430. SIP/2.0 200 OK
  431. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  432. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  433. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  434. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  435. CSeq: 19057 INVITE
  436. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  437. Supported: replaces
  438. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  439. Content-Type: application/sdp
  440. Content-Length: 273
  441.  
  442. v=0
  443. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  444. s=sipgate VoIP GW
  445. c=IN IP4 212.9.44.251
  446. t=0 0
  447. m=audio 22750 RTP/AVP 0 101
  448. a=rtpmap:0 PCMU/8000
  449. a=rtpmap:101 telephone-event/8000
  450. a=fmtp:101 0-16
  451. a=silenceSupp:off - - - -
  452. a=ptime:20
  453. a=sendrecv
  454. a=rtcp:22751
  455.  
  456. > 0x7f34440272b0 -- Strict RTP learning after remote address set to: 212.9.44.251:22750
  457. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  458. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  459. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjbb6b1380-bbe4-4406-9937-e76fbd62c2bd
  460. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  461. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  462. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  463. CSeq: 19057 ACK
  464. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  465. Route: <sip:172.20.40.5;lr>
  466. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  467. Max-Forwards: 70
  468. User-Agent: Asterisk PBX 15.0.0
  469. Content-Length: 0
  470.  
  471.  
  472. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  473. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  474. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  475. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  476. To: <sip:1002@10.0.0.12>
  477. Contact: <sip:1002@10.0.0.16:5060>
  478. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  479. CSeq: 53198 OPTIONS
  480. Max-Forwards: 70
  481. User-Agent: Asterisk PBX 15.0.0
  482. Content-Length: 0
  483.  
  484.  
  485. <--- Received SIP response (404 bytes) from UDP:10.0.0.30:5060 --->
  486. SIP/2.0 100 Trying
  487. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPjc2ba0de0-9fb2-4027-9773-56c6e453a5bc;received=10.0.0.16
  488. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  489. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  490. To: <sip:1001@10.0.0.30>;tag=1252544688
  491. CSeq: 14999 INVITE
  492. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  493. Content-Length: 0
  494.  
  495.  
  496. > 0x7f3444043da0 -- Strict RTP switching to RTP target address 10.0.0.30:16022 as source
  497. > 0x7f34440272b0 -- Strict RTP switching to RTP target address 212.9.44.251:22750 as source
  498. <--- Received SIP response (805 bytes) from UDP:10.0.0.30:5060 --->
  499. SIP/2.0 200 OK
  500. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPjc2ba0de0-9fb2-4027-9773-56c6e453a5bc;received=10.0.0.16
  501. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  502. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  503. To: <sip:1001@10.0.0.30>;tag=1252544688
  504. CSeq: 14999 INVITE
  505. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  506. Supported: 100rel,precondition
  507. Contact: <sip:1001@10.0.0.30:5060>
  508. Require: replaces
  509. Content-Type: application/sdp
  510. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  511. Content-Length: 208
  512.  
  513. v=0
  514. o=- 1509055335 1509055336 IN IP4 10.0.0.30
  515. s=-
  516. c=IN IP4 10.0.0.30
  517. t=0 0
  518. m=audio 16022 RTP/AVP 0 101
  519. a=rtpmap:0 PCMU/8000
  520. a=rtpmap:101 telephone-event/8000
  521. a=fmtp:101 0-15
  522. a=sendrecv
  523. a=ptime:20
  524.  
  525. <--- Transmitting SIP request (394 bytes) to UDP:10.0.0.30:5060 --->
  526. ACK sip:1001@10.0.0.30:5060 SIP/2.0
  527. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj83cfd54a-205b-4d22-91d0-f06f53b7a9ac
  528. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  529. To: <sip:1001@10.0.0.30>;tag=1252544688
  530. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  531. CSeq: 14999 ACK
  532. Max-Forwards: 70
  533. User-Agent: Asterisk PBX 15.0.0
  534. Content-Length: 0
  535.  
  536.  
  537. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  538. SIP/2.0 200 OK
  539. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  540. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  541. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  542. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  543. CSeq: 19057 INVITE
  544. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  545. Supported: replaces
  546. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  547. Content-Type: application/sdp
  548. Content-Length: 273
  549.  
  550. v=0
  551. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  552. s=sipgate VoIP GW
  553. c=IN IP4 212.9.44.251
  554. t=0 0
  555. m=audio 22750 RTP/AVP 0 101
  556. a=rtpmap:0 PCMU/8000
  557. a=rtpmap:101 telephone-event/8000
  558. a=fmtp:101 0-16
  559. a=silenceSupp:off - - - -
  560. a=ptime:20
  561. a=sendrecv
  562. a=rtcp:22751
  563.  
  564. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  565. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  566. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjbb6b1380-bbe4-4406-9937-e76fbd62c2bd
  567. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  568. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  569. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  570. CSeq: 19057 ACK
  571. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  572. Route: <sip:172.20.40.5;lr>
  573. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  574. Max-Forwards: 70
  575. User-Agent: Asterisk PBX 15.0.0
  576. Content-Length: 0
  577.  
  578.  
  579. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  580. SIP/2.0 200 OK
  581. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  582. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  583. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  584. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  585. CSeq: 19057 INVITE
  586. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  587. Supported: replaces
  588. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  589. Content-Type: application/sdp
  590. Content-Length: 273
  591.  
  592. v=0
  593. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  594. s=sipgate VoIP GW
  595. c=IN IP4 212.9.44.251
  596. t=0 0
  597. m=audio 22750 RTP/AVP 0 101
  598. a=rtpmap:0 PCMU/8000
  599. a=rtpmap:101 telephone-event/8000
  600. a=fmtp:101 0-16
  601. a=silenceSupp:off - - - -
  602. a=ptime:20
  603. a=sendrecv
  604. a=rtcp:22751
  605.  
  606. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  607. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  608. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjbb6b1380-bbe4-4406-9937-e76fbd62c2bd
  609. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  610. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  611. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  612. CSeq: 19057 ACK
  613. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  614. Route: <sip:172.20.40.5;lr>
  615. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  616. Max-Forwards: 70
  617. User-Agent: Asterisk PBX 15.0.0
  618. Content-Length: 0
  619.  
  620.  
  621. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  622. SIP/2.0 200 OK
  623. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  624. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  625. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  626. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  627. CSeq: 19057 INVITE
  628. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  629. Supported: replaces
  630. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  631. Content-Type: application/sdp
  632. Content-Length: 273
  633.  
  634. v=0
  635. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  636. s=sipgate VoIP GW
  637. c=IN IP4 212.9.44.251
  638. t=0 0
  639. m=audio 22750 RTP/AVP 0 101
  640. a=rtpmap:0 PCMU/8000
  641. a=rtpmap:101 telephone-event/8000
  642. a=fmtp:101 0-16
  643. a=silenceSupp:off - - - -
  644. a=ptime:20
  645. a=sendrecv
  646. a=rtcp:22751
  647.  
  648. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  649. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  650. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjbb6b1380-bbe4-4406-9937-e76fbd62c2bd
  651. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  652. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  653. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  654. CSeq: 19057 ACK
  655. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  656. Route: <sip:172.20.40.5;lr>
  657. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  658. Max-Forwards: 70
  659. User-Agent: Asterisk PBX 15.0.0
  660. Content-Length: 0
  661.  
  662.  
  663. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  664. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  665. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  666. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  667. To: <sip:1002@10.0.0.12>
  668. Contact: <sip:1002@10.0.0.16:5060>
  669. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  670. CSeq: 53198 OPTIONS
  671. Max-Forwards: 70
  672. User-Agent: Asterisk PBX 15.0.0
  673. Content-Length: 0
  674.  
  675.  
  676. <--- Transmitting SIP request (444 bytes) to UDP:217.10.68.151:5060 --->
  677. OPTIONS sip:sipconnect.sipgate.co.uk SIP/2.0
  678. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjb0e1bf8c-b567-4b2b-8575-7ca59171b954
  679. From: <sip:SIPID@sipconnect.sipgate.co.uk>;tag=2313b961-fcaa-40ce-ad52-24581a216ca6
  680. To: <sip:sipconnect.sipgate.co.uk>
  681. Contact: <sip:SIPID@10.0.0.16:5060>
  682. Call-ID: f8ba390c-65ea-48d3-a732-bffc95f2bb1e
  683. CSeq: 9549 OPTIONS
  684. Max-Forwards: 70
  685. User-Agent: Asterisk PBX 15.0.0
  686. Content-Length: 0
  687.  
  688.  
  689. <--- Received SIP response (457 bytes) from UDP:217.10.68.151:5060 --->
  690. SIP/2.0 200 OK
  691. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPjb0e1bf8c-b567-4b2b-8575-7ca59171b954;received=MYIP
  692. From: <sip:SIPID@sipconnect.sipgate.co.uk>;tag=2313b961-fcaa-40ce-ad52-24581a216ca6
  693. To: <sip:sipconnect.sipgate.co.uk>;tag=4c4f86a39a85efb0838c3d40da38cdad.c37f
  694. Call-ID: f8ba390c-65ea-48d3-a732-bffc95f2bb1e
  695. CSeq: 9549 OPTIONS
  696. Accept: */*
  697. Accept-Encoding:
  698. Accept-Language: en
  699. Supported:
  700. Content-Length: 0
  701.  
  702.  
  703. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  704. SIP/2.0 200 OK
  705. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  706. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  707. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  708. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  709. CSeq: 19057 INVITE
  710. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  711. Supported: replaces
  712. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  713. Content-Type: application/sdp
  714. Content-Length: 273
  715.  
  716. v=0
  717. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  718. s=sipgate VoIP GW
  719. c=IN IP4 212.9.44.251
  720. t=0 0
  721. m=audio 22750 RTP/AVP 0 101
  722. a=rtpmap:0 PCMU/8000
  723. a=rtpmap:101 telephone-event/8000
  724. a=fmtp:101 0-16
  725. a=silenceSupp:off - - - -
  726. a=ptime:20
  727. a=sendrecv
  728. a=rtcp:22751
  729.  
  730. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  731. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  732. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjbb6b1380-bbe4-4406-9937-e76fbd62c2bd
  733. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  734. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  735. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  736. CSeq: 19057 ACK
  737. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  738. Route: <sip:172.20.40.5;lr>
  739. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  740. Max-Forwards: 70
  741. User-Agent: Asterisk PBX 15.0.0
  742. Content-Length: 0
  743.  
  744.  
  745. <--- Received SIP request (835 bytes) from UDP:10.0.0.30:5060 --->
  746. INVITE sip:asterisk@10.0.0.16:5060 SIP/2.0
  747. Max-Breadth: 60
  748. Via: SIP/2.0/UDP 10.0.0.30:5060;branch=z9hG4bK72d104e7;rport
  749. Max-Forwards: 70
  750. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  751. From: <sip:1001@10.0.0.30>;tag=1252544688
  752. To: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  753. CSeq: 1 INVITE
  754. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  755. Contact: <sip:1001@10.0.0.30:5060>
  756. Content-Type: application/sdp
  757. User-Agent: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  758. Content-Length: 280
  759.  
  760. v=0
  761. o=- 1509055335 1509055337 IN IP4 10.0.0.30
  762. s=-
  763. c=IN IP4 0.0.0.0
  764. t=0 0
  765. m=audio 16022 RTP/AVP 9 8 18 0 101
  766. a=rtpmap:9 G722/8000
  767. a=rtpmap:8 PCMA/8000
  768. a=rtpmap:18 G729/8000
  769. a=rtpmap:0 PCMU/8000
  770. a=rtpmap:101 telephone-event/8000
  771. a=fmtp:101 0-15
  772. a=sendonly
  773. a=ptime:20
  774.  
  775. > 0x7f3444043da0 -- Strict RTP learning after remote address set to: 0.0.0.0:16022
  776. <--- Transmitting SIP response (814 bytes) to UDP:10.0.0.30:5060 --->
  777. SIP/2.0 200 OK
  778. Via: SIP/2.0/UDP 10.0.0.30:5060;rport=5060;received=10.0.0.30;branch=z9hG4bK72d104e7
  779. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  780. From: <sip:1001@10.0.0.30>;tag=1252544688
  781. To: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  782. CSeq: 1 INVITE
  783. Contact: <sip:asterisk@10.0.0.16:5060>
  784. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  785. Supported: 100rel, timer, replaces, norefersub
  786. Server: Asterisk PBX 15.0.0
  787. Content-Type: application/sdp
  788. Content-Length: 237
  789.  
  790. v=0
  791. o=- 1393429158 1393429160 IN IP4 212.9.44.251
  792. s=Asterisk
  793. c=IN IP4 212.9.44.251
  794. t=0 0
  795. m=audio 22750 RTP/AVP 0 101
  796. a=rtpmap:0 PCMU/8000
  797. a=rtpmap:101 telephone-event/8000
  798. a=fmtp:101 0-16
  799. a=ptime:20
  800. a=maxptime:150
  801. a=recvonly
  802.  
  803. -- Started music on hold, class 'default', on channel 'PJSIP/sipgate-00000006'
  804. <--- Transmitting SIP request (1117 bytes) to UDP:217.10.68.151:5060 --->
  805. INVITE sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  806. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj9075e5d2-185e-459d-a751-697c68f06f4c
  807. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  808. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  809. Contact: <sip:10.0.0.16:5060>
  810. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  811. CSeq: 19058 INVITE
  812. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  813. Route: <sip:172.20.40.5;lr>
  814. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  815. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  816. Supported: 100rel, timer, replaces, norefersub
  817. Session-Expires: 1800
  818. Min-SE: 90
  819. Max-Forwards: 70
  820. User-Agent: Asterisk PBX 15.0.0
  821. Content-Type: application/sdp
  822. Content-Length: 255
  823.  
  824. v=0
  825. o=- 1607684837 1607684842 IN IP4 10.0.0.16
  826. s=Asterisk
  827. c=IN IP4 10.0.0.16
  828. t=0 0
  829. m=audio 17714 RTP/AVP 0 8 101
  830. a=rtpmap:0 PCMU/8000
  831. a=rtpmap:8 PCMA/8000
  832. a=rtpmap:101 telephone-event/8000
  833. a=fmtp:101 0-16
  834. a=ptime:20
  835. a=maxptime:150
  836. a=sendrecv
  837.  
  838. <--- Received SIP response (497 bytes) from UDP:217.10.68.151:5060 --->
  839. SIP/2.0 100 trying -- your call is important to us
  840. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj9075e5d2-185e-459d-a751-697c68f06f4c
  841. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  842. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  843. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  844. CSeq: 19058 INVITE
  845. Content-Length: 0
  846. X-GeoIP: DE
  847. X-src_addr: 217.10.68.151:5060
  848.  
  849.  
  850. <--- Received SIP response (497 bytes) from UDP:217.10.68.151:5060 --->
  851. SIP/2.0 403 Forbidden (client configuration error)
  852. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj9075e5d2-185e-459d-a751-697c68f06f4c
  853. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  854. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  855. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  856. CSeq: 19058 INVITE
  857. Content-Length: 0
  858. X-GeoIP: DE
  859. X-src_addr: 217.10.68.151:5060
  860.  
  861.  
  862. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  863. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  864. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj9075e5d2-185e-459d-a751-697c68f06f4c
  865. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  866. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  867. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  868. CSeq: 19058 ACK
  869. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  870. Route: <sip:172.20.40.5;lr>
  871. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  872. Max-Forwards: 70
  873. User-Agent: Asterisk PBX 15.0.0
  874. Content-Length: 0
  875.  
  876.  
  877. <--- Transmitting SIP request (621 bytes) to UDP:217.10.68.151:5060 --->
  878. BYE sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  879. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj8cf1440f-b0d6-4702-bea7-814c2f696aed
  880. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  881. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  882. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  883. CSeq: 19059 BYE
  884. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  885. Route: <sip:172.20.40.5;lr>
  886. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  887. Reason: Q.850;cause=16
  888. Max-Forwards: 70
  889. User-Agent: Asterisk PBX 15.0.0
  890. Content-Length: 0
  891.  
  892.  
  893. <--- Received SIP request (403 bytes) from UDP:10.0.0.30:5060 --->
  894. ACK sip:asterisk@10.0.0.16:5060 SIP/2.0
  895. Max-Breadth: 60
  896. Via: SIP/2.0/UDP 10.0.0.30:5060;branch=z9hG4bK2da7e800;rport
  897. Max-Forwards: 70
  898. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  899. From: <sip:1001@10.0.0.30>;tag=1252544688
  900. To: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  901. CSeq: 1 ACK
  902. User-Agent: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  903. Content-Length: 0
  904.  
  905.  
  906. <--- Received SIP response (474 bytes) from UDP:217.10.68.151:5060 --->
  907. SIP/2.0 481 Call/Transaction does not exist
  908. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj8cf1440f-b0d6-4702-bea7-814c2f696aed
  909. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  910. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  911. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  912. CSeq: 19059 BYE
  913. Content-Length: 0
  914. X-src_addr: 217.10.68.151:5060
  915.  
  916.  
  917. -- Stopped music on hold on PJSIP/sipgate-00000006
  918. -- Channel PJSIP/sipgate-00000006 left 'native_rtp' basic-bridge <4b5c2154-0be9-4d56-a58b-3bc2e3b99d8a>
  919. -- Channel PJSIP/1001-00000007 left 'native_rtp' basic-bridge <4b5c2154-0be9-4d56-a58b-3bc2e3b99d8a>
  920. == Spawn extension (from-external, NUMBERTRUNK, 3) exited non-zero on 'PJSIP/sipgate-00000006'
  921. <--- Transmitting SIP request (1087 bytes) to UDP:10.0.0.30:5060 --->
  922. INVITE sip:1001@10.0.0.30:5060 SIP/2.0
  923. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj7648515d-5775-4b07-a2f8-20049133a797
  924. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  925. To: <sip:1001@10.0.0.30>;tag=1252544688
  926. Contact: <sip:asterisk@10.0.0.16:5060>
  927. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  928. CSeq: 15000 INVITE
  929. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  930. Supported: 100rel, timer, replaces, norefersub
  931. Session-Expires: 1800
  932. Min-SE: 90
  933. P-Asserted-Identity: "External" <sip:NUMBERMOBILE@10.0.0.16>
  934. Remote-Party-ID: "External" <sip:NUMBERMOBILE@10.0.0.16>;privacy=off;screen=no
  935. Max-Forwards: 70
  936. User-Agent: Asterisk PBX 15.0.0
  937. Content-Type: application/sdp
  938. Content-Length: 279
  939.  
  940. v=0
  941. o=- 1393429158 1393429161 IN IP4 10.0.0.16
  942. s=Asterisk
  943. c=IN IP4 10.0.0.16
  944. t=0 0
  945. m=audio 11542 RTP/AVP 9 0 8 101
  946. a=rtpmap:9 G722/8000
  947. a=rtpmap:0 PCMU/8000
  948. a=rtpmap:8 PCMA/8000
  949. a=rtpmap:101 telephone-event/8000
  950. a=fmtp:101 0-16
  951. a=ptime:20
  952. a=maxptime:150
  953. a=sendrecv
  954.  
  955. <--- Received SIP response (404 bytes) from UDP:10.0.0.30:5060 --->
  956. SIP/2.0 100 Trying
  957. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj7648515d-5775-4b07-a2f8-20049133a797;received=10.0.0.16
  958. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  959. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  960. To: <sip:1001@10.0.0.30>;tag=1252544688
  961. CSeq: 15000 INVITE
  962. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  963. Content-Length: 0
  964.  
  965.  
  966. <--- Received SIP response (803 bytes) from UDP:10.0.0.30:5060 --->
  967. SIP/2.0 200 OK
  968. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj7648515d-5775-4b07-a2f8-20049133a797;received=10.0.0.16
  969. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  970. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  971. To: <sip:1001@10.0.0.30>;tag=1252544688
  972. CSeq: 15000 INVITE
  973. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  974. Supported: 100rel,precondition
  975. Contact: <sip:1001@10.0.0.30:5060>
  976. Require: replaces
  977. Content-Type: application/sdp
  978. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  979. Content-Length: 206
  980.  
  981. v=0
  982. o=- 1509055335 1509055338 IN IP4 10.0.0.30
  983. s=-
  984. c=IN IP4 0.0.0.0
  985. t=0 0
  986. m=audio 16022 RTP/AVP 9 101
  987. a=rtpmap:9 G722/8000
  988. a=rtpmap:101 telephone-event/8000
  989. a=fmtp:101 0-15
  990. a=inactive
  991. a=ptime:20
  992.  
  993. <--- Transmitting SIP request (394 bytes) to UDP:10.0.0.30:5060 --->
  994. ACK sip:1001@10.0.0.30:5060 SIP/2.0
  995. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj66136176-19d6-4589-afd2-3d80e1ba35be
  996. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  997. To: <sip:1001@10.0.0.30>;tag=1252544688
  998. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  999. CSeq: 15000 ACK
  1000. Max-Forwards: 70
  1001. User-Agent: Asterisk PBX 15.0.0
  1002. Content-Length: 0
  1003.  
  1004.  
  1005. <--- Transmitting SIP request (394 bytes) to UDP:10.0.0.30:5060 --->
  1006. BYE sip:1001@10.0.0.30:5060 SIP/2.0
  1007. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj5e45b520-4e6f-4cc2-b60f-40dde2aa09e0
  1008. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  1009. To: <sip:1001@10.0.0.30>;tag=1252544688
  1010. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  1011. CSeq: 15001 BYE
  1012. Max-Forwards: 70
  1013. User-Agent: Asterisk PBX 15.0.0
  1014. Content-Length: 0
  1015.  
  1016.  
  1017. <--- Received SIP response (474 bytes) from UDP:10.0.0.30:5060 --->
  1018. SIP/2.0 200 OK
  1019. Via: SIP/2.0/UDP 10.0.0.16:5060;rport=5060;branch=z9hG4bKPj5e45b520-4e6f-4cc2-b60f-40dde2aa09e0;received=10.0.0.16
  1020. Call-ID: 0cebbdb8-a3b9-443e-bd10-d63589c00aa4
  1021. From: "External" <sip:NUMBERMOBILE@10.0.0.16>;tag=b2ec4b29-244a-4cdd-9336-d7aa92ac0028
  1022. To: <sip:1001@10.0.0.30>;tag=1252544688
  1023. CSeq: 15001 BYE
  1024. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER
  1025. Server: Panasonic-KX-HDV230X/06.101 (bcc3424bb637)
  1026. Content-Length: 0
  1027.  
  1028.  
  1029. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  1030. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  1031. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  1032. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  1033. To: <sip:1002@10.0.0.12>
  1034. Contact: <sip:1002@10.0.0.16:5060>
  1035. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  1036. CSeq: 53198 OPTIONS
  1037. Max-Forwards: 70
  1038. User-Agent: Asterisk PBX 15.0.0
  1039. Content-Length: 0
  1040.  
  1041.  
  1042. <--- Received SIP response (873 bytes) from UDP:217.10.68.151:5060 --->
  1043. SIP/2.0 200 OK
  1044. Via: SIP/2.0/UDP 10.0.0.16:5060;received=MYIP;rport=5060;branch=z9hG4bKPj885e16bb-798b-4b20-9e2f-a0af270e8e27
  1045. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  1046. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  1047. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  1048. CSeq: 19057 INVITE
  1049. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  1050. Supported: replaces
  1051. Contact: <sip:NUMBERMOBILE@217.116.117.7:5060>
  1052. Content-Type: application/sdp
  1053. Content-Length: 273
  1054.  
  1055. v=0
  1056. o=root 1607684837 1607684839 IN IP4 217.116.117.7
  1057. s=sipgate VoIP GW
  1058. c=IN IP4 212.9.44.251
  1059. t=0 0
  1060. m=audio 22750 RTP/AVP 0 101
  1061. a=rtpmap:0 PCMU/8000
  1062. a=rtpmap:101 telephone-event/8000
  1063. a=fmtp:101 0-16
  1064. a=silenceSupp:off - - - -
  1065. a=ptime:20
  1066. a=sendrecv
  1067. a=rtcp:22751
  1068.  
  1069. <--- Transmitting SIP request (597 bytes) to UDP:217.10.68.151:5060 --->
  1070. ACK sip:NUMBERMOBILE@217.10.68.151:5060 SIP/2.0
  1071. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPjc1b68b93-943c-40ab-8706-0779a4262510
  1072. From: <sip:00NUMBERTRUNK@sipconnect.sipgate.co.uk>;tag=9b913695-4241-4d6c-8454-aca110e6bdd1
  1073. To: "NUMBERMOBILE" <sip:NUMBERMOBILE@sipconnect.sipgate.co.uk>;tag=as1bc1d144
  1074. Call-ID: 727c6b8f21c1d9f670db98e6488829ee@sipconnect.sipgate.co.uk
  1075. CSeq: 19057 ACK
  1076. Route: <sip:217.10.68.151:5060;lr;ftag=as1bc1d144>
  1077. Route: <sip:172.20.40.5;lr>
  1078. Route: <sip:217.10.68.137;lr;ftag=as1bc1d144>
  1079. Max-Forwards: 70
  1080. User-Agent: Asterisk PBX 15.0.0
  1081. Content-Length: 0
  1082.  
  1083.  
  1084. <--- Transmitting SIP request (405 bytes) to UDP:10.0.0.12:5060 --->
  1085. OPTIONS sip:1002@10.0.0.12:5060 SIP/2.0
  1086. Via: SIP/2.0/UDP 10.0.0.16:5060;rport;branch=z9hG4bKPj154f40f6-e660-46e3-835d-6e6874d428f7
  1087. From: <sip:1002@10.0.0.16>;tag=3930737c-67b7-41ec-8be9-2a7e63a74ea4
  1088. To: <sip:1002@10.0.0.12>
  1089. Contact: <sip:1002@10.0.0.16:5060>
  1090. Call-ID: f0a8878d-d445-4ed7-b129-3ba023e6b3cf
  1091. CSeq: 53198 OPTIONS
  1092. Max-Forwards: 70
  1093. User-Agent: Asterisk PBX 15.0.0
  1094. Content-Length: 0
  1095.  
  1096.  
  1097. asterisk*CLI> pjsip set logger off
  1098. PJSIP Logging disabled
  1099. asterisk*CLI>
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