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Asterisk Debug

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Jul 28th, 2015
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  1. Retransmitting #8 (NAT) to 23.92.94.65:5071:
  2. SIP/2.0 401 Unauthorized
  3. Via: SIP/2.0/UDP 23.92.94.65:5071;branch=z9hG4bK-d3c7fc9869d505379c333a99b41eb7d e;received=23.92.94.65;rport=5071
  4. From: 50001<sip:50001@198.245.62.16>;tag=485c4dfb
  5. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as517188fe
  6. Call-ID: d3c7fc9869d505379c333a99b41eb7de
  7. CSeq: 1 INVITE
  8. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  10. Supported: replaces
  11. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23670c41"
  12. Content-Length: 0
  13.  
  14.  
  15. ---
  16. Retransmitting #9 (NAT) to 23.92.94.65:5071:
  17. SIP/2.0 401 Unauthorized
  18. Via: SIP/2.0/UDP 23.92.94.65:5071;branch=z9hG4bK-d3c7fc9869d505379c333a99b41eb7d e;received=23.92.94.65;rport=5071
  19. From: 50001<sip:50001@198.245.62.16>;tag=485c4dfb
  20. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as517188fe
  21. Call-ID: d3c7fc9869d505379c333a99b41eb7de
  22. CSeq: 1 INVITE
  23. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  24. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  25. Supported: replaces
  26. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23670c41"
  27. Content-Length: 0
  28.  
  29.  
  30. ---
  31. Retransmitting #10 (NAT) to 23.92.94.65:5071:
  32. SIP/2.0 401 Unauthorized
  33. Via: SIP/2.0/UDP 23.92.94.65:5071;branch=z9hG4bK-d3c7fc9869d505379c333a99b41eb7d e;received=23.92.94.65;rport=5071
  34. From: 50001<sip:50001@198.245.62.16>;tag=485c4dfb
  35. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as517188fe
  36. Call-ID: d3c7fc9869d505379c333a99b41eb7de
  37. CSeq: 1 INVITE
  38. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  39. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  40. Supported: replaces
  41. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23670c41"
  42. Content-Length: 0
  43.  
  44.  
  45. ---
  46. Really destroying SIP dialog '95f5da82-11ab32f-5b60852@216.115.69.131' Method: O PTIONS
  47. [Jul 28 13:40:30] WARNING[23666]: chan_sip.c:4175 retrans_pkt: Retransmission ti meout reached on transmission d3c7fc9869d505379c333a99b41eb7de for seqno 1 (Non- critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retrans missions
  48. Packet timed out after 31999ms with no response
  49. Really destroying SIP dialog 'd3c7fc9869d505379c333a99b41eb7de' Method: INVITE
  50. Really destroying SIP dialog '0c26e475-aeb28168-14b900a@70.167.153.136' Method: OPTIONS
  51.  
  52. <--- SIP read from UDP:216.115.69.144:5060 --->
  53. OPTIONS sip:198.245.62.16:5060 SIP/2.0
  54. Max-Forwards: 10
  55. Record-Route: <sip:216.115.69.144;lr>
  56. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e87.a8dbc1816846d5945d589686fbffa af9.0
  57. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  58. Route: <sip:216.115.69.144;lr;received='sip:198.245.62.16:5060'>
  59. From: sip:ping@invalid;tag=095dd61f
  60. To: sip:198.245.62.16:5060
  61. Call-ID: 95f5da82-6ddd32f-ce60852@216.115.69.131
  62. CSeq: 1 OPTIONS
  63. Content-Length: 0
  64.  
  65. <------------->
  66. --- (11 headers 0 lines) ---
  67. Sending to 216.115.69.144:5060 (NAT)
  68. Looking for s in internal (domain 198.245.62.16)
  69.  
  70. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  71. SIP/2.0 200 OK
  72. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e87.a8dbc1816846d5945d589686fbffa af9.0;received=216.115.69.144;rport=5060
  73. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  74. Record-Route: <sip:216.115.69.144;lr>
  75. From: sip:ping@invalid;tag=095dd61f
  76. To: sip:198.245.62.16:5060;tag=as687fad8f
  77. Call-ID: 95f5da82-6ddd32f-ce60852@216.115.69.131
  78. CSeq: 1 OPTIONS
  79. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  80. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  81. Supported: replaces
  82. Contact: <sip:127.0.0.1:5060>
  83. Accept: application/sdp
  84. Content-Length: 0
  85.  
  86.  
  87. <------------>
  88. Scheduling destruction of SIP dialog '95f5da82-6ddd32f-ce60852@216.115.69.131' i n 32000 ms (Method: OPTIONS)
  89.  
  90. <--- SIP read from UDP:162.197.74.233:53103 --->
  91. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  92. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---b9482474b673b3c7 ;rport
  93. Max-Forwards: 70
  94. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>
  95. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  96. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  97. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  98. CSeq: 1 REGISTER
  99. Expires: 60
  100. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  101. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  102. User-Agent: Zoiper r30645
  103. Allow-Events: presence, kpml
  104. Content-Length: 0
  105.  
  106. <------------->
  107. --- (14 headers 0 lines) ---
  108. Sending to 162.197.74.233:53103 (NAT)
  109. Sending to 162.197.74.233:53103 (NAT)
  110.  
  111. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  112. SIP/2.0 401 Unauthorized
  113. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---b9482474b673b3c7 ;received=162.197.74.233;rport=53103
  114. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  115. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as4ae03d96
  116. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  117. CSeq: 1 REGISTER
  118. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  119. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  120. Supported: replaces
  121. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00c9d74b"
  122. Content-Length: 0
  123.  
  124.  
  125. <------------>
  126. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Met hod: REGISTER)
  127.  
  128. <--- SIP read from UDP:162.197.74.233:53103 --->
  129. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  130. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---4a59d894d4bbfc6a ;rport
  131. Max-Forwards: 70
  132. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>
  133. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  134. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  135. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  136. CSeq: 2 REGISTER
  137. Expires: 60
  138. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  139. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  140. User-Agent: Zoiper r30645
  141. Authorization: Digest username="12345424534",realm="asterisk",nonce="00c9d74b",u ri="sip:198.245.62.16:5060;transport=UDP",response="b55c8131433b5bb93bcbb2f86bf3 9ff3",algorithm=MD5
  142. Allow-Events: presence, kpml
  143. Content-Length: 0
  144.  
  145. <------------->
  146. --- (15 headers 0 lines) ---
  147. Sending to 162.197.74.233:53103 (NAT)
  148.  
  149. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  150. SIP/2.0 200 OK
  151. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---4a59d894d4bbfc6a ;received=162.197.74.233;rport=53103
  152. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  153. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as4ae03d96
  154. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  155. CSeq: 2 REGISTER
  156. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  157. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  158. Supported: replaces
  159. Expires: 60
  160. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>;expires=60
  161. Date: Tue, 28 Jul 2015 17:40:58 GMT
  162. Content-Length: 0
  163.  
  164.  
  165. <------------>
  166. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Met hod: REGISTER)
  167. [Jul 28 13:41:09] NOTICE[23666]: chan_sip.c:15142 sip_reregister: -- Re-regis tration for 68511092@sip.flowroute.com
  168. REGISTER 11 headers, 0 lines
  169. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  170. REGISTER sip:sip.flowroute.com SIP/2.0
  171. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2046bc6a;rport
  172. Max-Forwards: 70
  173. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  174. To: <sip:68511092@sip.flowroute.com>
  175. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  176. CSeq: 653 REGISTER
  177. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  178. Authorization: Digest username="68511092", realm="sip.flowroute.com", algorithm= MD5, uri="sip:sip.flowroute.com", nonce="Vbe/FFW3veiMzl1QiinyvfKFw+jIhuGh", resp onse="4d9006801aebc42eba52902a82c8b130", qop=auth, cnonce="7149dbf7", nc=0000000 3
  179. Expires: 120
  180. Contact: <sip:s@127.0.0.1:5060>
  181. Content-Length: 0
  182.  
  183.  
  184. ---
  185.  
  186. <--- SIP read from UDP:216.115.69.144:5060 --->
  187. SIP/2.0 200 OK
  188. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK2046bc6a;rp ort=5060
  189. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  190. To: <sip:68511092@sip.flowroute.com>;tag=aa681f9fdf30149b00040f579a1d99c4.df02
  191. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  192. CSeq: 653 REGISTER
  193. Contact: <sip:s@127.0.0.1:5060>;q=1;expires=120;received="sip:198.245.62.16:5060 "
  194. Content-Length: 0
  195.  
  196. <------------->
  197. --- (8 headers 0 lines) ---
  198. [Jul 28 13:41:09] NOTICE[23666]: chan_sip.c:23655 handle_response_register: Outb ound Registration: Expiry for sip.flowroute.com is 120 sec (Scheduling reregistr ation in 105 s)
  199. Really destroying SIP dialog '02bb0a1d18d326ac35294d9425595a75@[::1]' Method: RE GISTER
  200.  
  201. <--- SIP read from UDP:216.115.69.144:5060 --->
  202. OPTIONS sip:198.245.62.16:5060 SIP/2.0
  203. Max-Forwards: 10
  204. Record-Route: <sip:216.115.69.144;lr>
  205. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK9df4.8968f4a2d8f16f53d3e8005f5cfd7 f56.0
  206. Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
  207. Route: <sip:216.115.69.144;lr;received='sip:198.245.62.16:5060'>
  208. From: sip:ping@invalid;tag=84d15d59
  209. To: sip:198.245.62.16:5060
  210. Call-ID: 0c26e475-eaf48168-97b900a@70.167.153.136
  211. CSeq: 1 OPTIONS
  212. Content-Length: 0
  213.  
  214. <------------->
  215. --- (11 headers 0 lines) ---
  216. Sending to 216.115.69.144:5060 (NAT)
  217. Looking for s in internal (domain 198.245.62.16)
  218.  
  219. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  220. SIP/2.0 200 OK
  221. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK9df4.8968f4a2d8f16f53d3e8005f5cfd7 f56.0;received=216.115.69.144;rport=5060
  222. Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
  223. Record-Route: <sip:216.115.69.144;lr>
  224. From: sip:ping@invalid;tag=84d15d59
  225. To: sip:198.245.62.16:5060;tag=as57151a5d
  226. Call-ID: 0c26e475-eaf48168-97b900a@70.167.153.136
  227. CSeq: 1 OPTIONS
  228. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  230. Supported: replaces
  231. Contact: <sip:127.0.0.1:5060>
  232. Accept: application/sdp
  233. Content-Length: 0
  234.  
  235.  
  236. <------------>
  237. Scheduling destruction of SIP dialog '0c26e475-eaf48168-97b900a@70.167.153.136' in 32000 ms (Method: OPTIONS)
  238. Really destroying SIP dialog '95f5da82-6ddd32f-ce60852@216.115.69.131' Method: O PTIONS
  239.  
  240. <--- SIP read from UDP:216.115.69.144:5060 --->
  241. INVITE sip:12345424534@thegamingcorner.net:5060 SIP/2.0
  242. Record-Route: <sip:216.115.69.144;lr>
  243. Max-Forwards: 66
  244. Record-Route: <sip:216.115.69.132;lr>
  245. To: <sip:+12345424534@fl.gg>
  246. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  247. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0
  248. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  249. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  250. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  251. Call-ID: 537699074_101142141@4.55.17.35
  252. CSeq: 5324 INVITE
  253. Contact: "DUFFIELD DAVID" <sip:+13305344879@4.55.17.35:5060>
  254. Content-Length: 215
  255. Content-Type: application/sdp
  256. P-Asserted-Identity: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>
  257.  
  258. v=0
  259. o=- 6095 10129 IN IP4 4.55.17.2
  260. s=-
  261. c=IN IP4 4.55.17.2
  262. t=0 0
  263. m=audio 17590 RTP/AVP 0 8 18 101
  264. a=rtpmap:18 G729/8000
  265. a=fmtp:18 annexb=no
  266. a=rtpmap:101 telephone-event/8000
  267. a=fmtp:101 0-15
  268. a=maxptime:20
  269. <------------->
  270. --- (16 headers 11 lines) ---
  271. Sending to 216.115.69.144:5060 (NAT)
  272. Sending to 216.115.69.144:5060 (NAT)
  273. Using INVITE request as basis request - 537699074_101142141@4.55.17.35
  274. Found peer 'flowroute' for '+13305344879' from 216.115.69.144:5060
  275. Found RTP audio format 0
  276. Found RTP audio format 8
  277. Found RTP audio format 18
  278. Found RTP audio format 101
  279. Found audio description format G729 for ID 18
  280. Found audio description format telephone-event for ID 101
  281. Capabilities: us - (ulaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/te xt=(nothing), combined - (ulaw|g729)
  282. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
  283. Peer audio RTP is at port 4.55.17.2:17590
  284. Looking for 12345424534 in inbound (domain thegamingcorner.net)
  285. list_route: hop: <sip:216.115.69.144;lr>
  286. list_route: hop: <sip:216.115.69.132;lr>
  287.  
  288. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  289. SIP/2.0 100 Trying
  290. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  291. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  292. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  293. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  294. Record-Route: <sip:216.115.69.144;lr>
  295. Record-Route: <sip:216.115.69.132;lr>
  296. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  297. To: <sip:+12345424534@fl.gg>
  298. Call-ID: 537699074_101142141@4.55.17.35
  299. CSeq: 5324 INVITE
  300. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  301. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  302. Supported: replaces
  303. Contact: <sip:12345424534@127.0.0.1:5060>
  304. Content-Length: 0
  305.  
  306.  
  307. <------------>
  308. Audio is at 17644
  309. Adding codec 100003 (ulaw) to SDP
  310. Adding codec 100008 (g729) to SDP
  311. Adding non-codec 0x1 (telephone-event) to SDP
  312.  
  313. <--- Reliably Transmitting (NAT) to 216.115.69.144:5060 --->
  314. SIP/2.0 200 OK
  315. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  316. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  317. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  318. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  319. Record-Route: <sip:216.115.69.144;lr>
  320. Record-Route: <sip:216.115.69.132;lr>
  321. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  322. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  323. Call-ID: 537699074_101142141@4.55.17.35
  324. CSeq: 5324 INVITE
  325. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  326. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  327. Supported: replaces
  328. Contact: <sip:12345424534@127.0.0.1:5060>
  329. Content-Type: application/sdp
  330. Content-Length: 289
  331.  
  332. v=0
  333. o=root 913076305 913076305 IN IP4 127.0.0.1
  334. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  335. c=IN IP4 127.0.0.1
  336. t=0 0
  337. m=audio 17644 RTP/AVP 0 18 101
  338. a=rtpmap:0 PCMU/8000
  339. a=rtpmap:18 G729/8000
  340. a=fmtp:18 annexb=no
  341. a=rtpmap:101 telephone-event/8000
  342. a=fmtp:101 0-16
  343. a=ptime:20
  344. a=sendrecv
  345.  
  346. <------------>
  347.  
  348. <--- SIP read from UDP:162.197.74.233:53103 --->
  349.  
  350.  
  351. <------------->
  352. Retransmitting #1 (NAT) to 216.115.69.144:5060:
  353. SIP/2.0 200 OK
  354. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  355. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  356. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  357. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  358. Record-Route: <sip:216.115.69.144;lr>
  359. Record-Route: <sip:216.115.69.132;lr>
  360. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  361. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  362. Call-ID: 537699074_101142141@4.55.17.35
  363. CSeq: 5324 INVITE
  364. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  365. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  366. Supported: replaces
  367. Contact: <sip:12345424534@127.0.0.1:5060>
  368. Content-Type: application/sdp
  369. Content-Length: 289
  370.  
  371. v=0
  372. o=root 913076305 913076305 IN IP4 127.0.0.1
  373. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  374. c=IN IP4 127.0.0.1
  375. t=0 0
  376. m=audio 17644 RTP/AVP 0 18 101
  377. a=rtpmap:0 PCMU/8000
  378. a=rtpmap:18 G729/8000
  379. a=fmtp:18 annexb=no
  380. a=rtpmap:101 telephone-event/8000
  381. a=fmtp:101 0-16
  382. a=ptime:20
  383. a=sendrecv
  384.  
  385. ---
  386. Scheduling destruction of SIP dialog '537699074_101142141@4.55.17.35' in 32000 m s (Method: INVITE)
  387. Retransmitting #2 (NAT) to 216.115.69.144:5060:
  388. SIP/2.0 200 OK
  389. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  390. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  391. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  392. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  393. Record-Route: <sip:216.115.69.144;lr>
  394. Record-Route: <sip:216.115.69.132;lr>
  395. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  396. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  397. Call-ID: 537699074_101142141@4.55.17.35
  398. CSeq: 5324 INVITE
  399. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  400. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  401. Supported: replaces
  402. Contact: <sip:12345424534@127.0.0.1:5060>
  403. Content-Type: application/sdp
  404. Content-Length: 289
  405.  
  406. v=0
  407. o=root 913076305 913076305 IN IP4 127.0.0.1
  408. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  409. c=IN IP4 127.0.0.1
  410. t=0 0
  411. m=audio 17644 RTP/AVP 0 18 101
  412. a=rtpmap:0 PCMU/8000
  413. a=rtpmap:18 G729/8000
  414. a=fmtp:18 annexb=no
  415. a=rtpmap:101 telephone-event/8000
  416. a=fmtp:101 0-16
  417. a=ptime:20
  418. a=sendrecv
  419.  
  420. ---
  421. Really destroying SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' Method: REGISTER
  422. Retransmitting #3 (NAT) to 216.115.69.144:5060:
  423. SIP/2.0 200 OK
  424. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  425. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  426. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  427. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  428. Record-Route: <sip:216.115.69.144;lr>
  429. Record-Route: <sip:216.115.69.132;lr>
  430. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  431. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  432. Call-ID: 537699074_101142141@4.55.17.35
  433. CSeq: 5324 INVITE
  434. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  435. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  436. Supported: replaces
  437. Contact: <sip:12345424534@127.0.0.1:5060>
  438. Content-Type: application/sdp
  439. Content-Length: 289
  440.  
  441. v=0
  442. o=root 913076305 913076305 IN IP4 127.0.0.1
  443. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  444. c=IN IP4 127.0.0.1
  445. t=0 0
  446. m=audio 17644 RTP/AVP 0 18 101
  447. a=rtpmap:0 PCMU/8000
  448. a=rtpmap:18 G729/8000
  449. a=fmtp:18 annexb=no
  450. a=rtpmap:101 telephone-event/8000
  451. a=fmtp:101 0-16
  452. a=ptime:20
  453. a=sendrecv
  454.  
  455. ---
  456. Retransmitting #4 (NAT) to 216.115.69.144:5060:
  457. SIP/2.0 200 OK
  458. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  459. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  460. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  461. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  462. Record-Route: <sip:216.115.69.144;lr>
  463. Record-Route: <sip:216.115.69.132;lr>
  464. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  465. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  466. Call-ID: 537699074_101142141@4.55.17.35
  467. CSeq: 5324 INVITE
  468. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  469. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  470. Supported: replaces
  471. Contact: <sip:12345424534@127.0.0.1:5060>
  472. Content-Type: application/sdp
  473. Content-Length: 289
  474.  
  475. v=0
  476. o=root 913076305 913076305 IN IP4 127.0.0.1
  477. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  478. c=IN IP4 127.0.0.1
  479. t=0 0
  480. m=audio 17644 RTP/AVP 0 18 101
  481. a=rtpmap:0 PCMU/8000
  482. a=rtpmap:18 G729/8000
  483. a=fmtp:18 annexb=no
  484. a=rtpmap:101 telephone-event/8000
  485. a=fmtp:101 0-16
  486. a=ptime:20
  487. a=sendrecv
  488.  
  489. ---
  490. Retransmitting #5 (NAT) to 216.115.69.144:5060:
  491. SIP/2.0 200 OK
  492. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  493. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  494. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  495. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  496. Record-Route: <sip:216.115.69.144;lr>
  497. Record-Route: <sip:216.115.69.132;lr>
  498. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  499. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  500. Call-ID: 537699074_101142141@4.55.17.35
  501. CSeq: 5324 INVITE
  502. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  503. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  504. Supported: replaces
  505. Contact: <sip:12345424534@127.0.0.1:5060>
  506. Content-Type: application/sdp
  507. Content-Length: 289
  508.  
  509. v=0
  510. o=root 913076305 913076305 IN IP4 127.0.0.1
  511. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  512. c=IN IP4 127.0.0.1
  513. t=0 0
  514. m=audio 17644 RTP/AVP 0 18 101
  515. a=rtpmap:0 PCMU/8000
  516. a=rtpmap:18 G729/8000
  517. a=fmtp:18 annexb=no
  518. a=rtpmap:101 telephone-event/8000
  519. a=fmtp:101 0-16
  520. a=ptime:20
  521. a=sendrecv
  522.  
  523. ---
  524. Retransmitting #6 (NAT) to 216.115.69.144:5060:
  525. SIP/2.0 200 OK
  526. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  527. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  528. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  529. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  530. Record-Route: <sip:216.115.69.144;lr>
  531. Record-Route: <sip:216.115.69.132;lr>
  532. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  533. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  534. Call-ID: 537699074_101142141@4.55.17.35
  535. CSeq: 5324 INVITE
  536. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  537. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  538. Supported: replaces
  539. Contact: <sip:12345424534@127.0.0.1:5060>
  540. Content-Type: application/sdp
  541. Content-Length: 289
  542.  
  543. v=0
  544. o=root 913076305 913076305 IN IP4 127.0.0.1
  545. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  546. c=IN IP4 127.0.0.1
  547. t=0 0
  548. m=audio 17644 RTP/AVP 0 18 101
  549. a=rtpmap:0 PCMU/8000
  550. a=rtpmap:18 G729/8000
  551. a=fmtp:18 annexb=no
  552. a=rtpmap:101 telephone-event/8000
  553. a=fmtp:101 0-16
  554. a=ptime:20
  555. a=sendrecv
  556.  
  557. ---
  558. Really destroying SIP dialog '0c26e475-eaf48168-97b900a@70.167.153.136' Method: OPTIONS
  559. Retransmitting #7 (NAT) to 216.115.69.144:5060:
  560. SIP/2.0 200 OK
  561. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  562. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  563. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  564. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  565. Record-Route: <sip:216.115.69.144;lr>
  566. Record-Route: <sip:216.115.69.132;lr>
  567. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  568. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  569. Call-ID: 537699074_101142141@4.55.17.35
  570. CSeq: 5324 INVITE
  571. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  572. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  573. Supported: replaces
  574. Contact: <sip:12345424534@127.0.0.1:5060>
  575. Content-Type: application/sdp
  576. Content-Length: 289
  577.  
  578. v=0
  579. o=root 913076305 913076305 IN IP4 127.0.0.1
  580. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  581. c=IN IP4 127.0.0.1
  582. t=0 0
  583. m=audio 17644 RTP/AVP 0 18 101
  584. a=rtpmap:0 PCMU/8000
  585. a=rtpmap:18 G729/8000
  586. a=fmtp:18 annexb=no
  587. a=rtpmap:101 telephone-event/8000
  588. a=fmtp:101 0-16
  589. a=ptime:20
  590. a=sendrecv
  591.  
  592. ---
  593.  
  594. <--- SIP read from UDP:216.115.69.144:5060 --->
  595. OPTIONS sip:198.245.62.16:5060 SIP/2.0
  596. Max-Forwards: 10
  597. Record-Route: <sip:216.115.69.144;lr>
  598. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK694b.495906329bb2e1a89701668ab4416 a25.0
  599. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  600. Route: <sip:216.115.69.144;lr;received='sip:198.245.62.16:5060'>
  601. From: sip:ping@invalid;tag=459fd61f
  602. To: sip:198.245.62.16:5060
  603. Call-ID: 95f5da82-a91042f-3270852@216.115.69.131
  604. CSeq: 1 OPTIONS
  605. Content-Length: 0
  606.  
  607. <------------->
  608. --- (11 headers 0 lines) ---
  609. Sending to 216.115.69.144:5060 (NAT)
  610. Looking for s in internal (domain 198.245.62.16)
  611.  
  612. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  613. SIP/2.0 200 OK
  614. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK694b.495906329bb2e1a89701668ab4416 a25.0;received=216.115.69.144;rport=5060
  615. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  616. Record-Route: <sip:216.115.69.144;lr>
  617. From: sip:ping@invalid;tag=459fd61f
  618. To: sip:198.245.62.16:5060;tag=as14920185
  619. Call-ID: 95f5da82-a91042f-3270852@216.115.69.131
  620. CSeq: 1 OPTIONS
  621. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  622. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  623. Supported: replaces
  624. Contact: <sip:127.0.0.1:5060>
  625. Accept: application/sdp
  626. Content-Length: 0
  627.  
  628.  
  629. <------------>
  630. Scheduling destruction of SIP dialog '95f5da82-a91042f-3270852@216.115.69.131' i n 32000 ms (Method: OPTIONS)
  631. Retransmitting #8 (NAT) to 216.115.69.144:5060:
  632. SIP/2.0 200 OK
  633. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  634. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  635. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  636. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  637. Record-Route: <sip:216.115.69.144;lr>
  638. Record-Route: <sip:216.115.69.132;lr>
  639. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  640. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  641. Call-ID: 537699074_101142141@4.55.17.35
  642. CSeq: 5324 INVITE
  643. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  644. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  645. Supported: replaces
  646. Contact: <sip:12345424534@127.0.0.1:5060>
  647. Content-Type: application/sdp
  648. Content-Length: 289
  649.  
  650. v=0
  651. o=root 913076305 913076305 IN IP4 127.0.0.1
  652. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  653. c=IN IP4 127.0.0.1
  654. t=0 0
  655. m=audio 17644 RTP/AVP 0 18 101
  656. a=rtpmap:0 PCMU/8000
  657. a=rtpmap:18 G729/8000
  658. a=fmtp:18 annexb=no
  659. a=rtpmap:101 telephone-event/8000
  660. a=fmtp:101 0-16
  661. a=ptime:20
  662. a=sendrecv
  663.  
  664. ---
  665.  
  666. <--- SIP read from UDP:162.197.74.233:53103 --->
  667. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  668. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---1225fa25383acb11 ;rport
  669. Max-Forwards: 70
  670. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>
  671. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  672. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  673. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  674. CSeq: 3 REGISTER
  675. Expires: 60
  676. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  677. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  678. User-Agent: Zoiper r30645
  679. Authorization: Digest username="12345424534",realm="asterisk",nonce="00c9d74b",u ri="sip:198.245.62.16:5060;transport=UDP",response="b55c8131433b5bb93bcbb2f86bf3 9ff3",algorithm=MD5
  680. Allow-Events: presence, kpml
  681. Content-Length: 0
  682.  
  683. <------------->
  684. --- (15 headers 0 lines) ---
  685. Sending to 162.197.74.233:53103 (NAT)
  686. Sending to 162.197.74.233:53103 (NAT)
  687.  
  688. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  689. SIP/2.0 401 Unauthorized
  690. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---1225fa25383acb11 ;received=162.197.74.233;rport=53103
  691. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  692. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as064003c8
  693. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  694. CSeq: 3 REGISTER
  695. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  696. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  697. Supported: replaces
  698. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3852a490"
  699. Content-Length: 0
  700.  
  701.  
  702. <------------>
  703. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Met hod: REGISTER)
  704.  
  705. <--- SIP read from UDP:162.197.74.233:53103 --->
  706. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  707. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---835e489d5e9cb2b3 ;rport
  708. Max-Forwards: 70
  709. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>
  710. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  711. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  712. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  713. CSeq: 4 REGISTER
  714. Expires: 60
  715. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  716. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  717. User-Agent: Zoiper r30645
  718. Authorization: Digest username="12345424534",realm="asterisk",nonce="3852a490",u ri="sip:198.245.62.16:5060;transport=UDP",response="c99faeaf0c4c2938f01674aff06f 8520",algorithm=MD5
  719. Allow-Events: presence, kpml
  720. Content-Length: 0
  721.  
  722. <------------->
  723. --- (15 headers 0 lines) ---
  724. Sending to 162.197.74.233:53103 (NAT)
  725.  
  726. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  727. SIP/2.0 200 OK
  728. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---835e489d5e9cb2b3 ;received=162.197.74.233;rport=53103
  729. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  730. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as064003c8
  731. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  732. CSeq: 4 REGISTER
  733. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  734. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  735. Supported: replaces
  736. Expires: 60
  737. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transp ort=UDP>;expires=60
  738. Date: Tue, 28 Jul 2015 17:41:52 GMT
  739. Content-Length: 0
  740.  
  741.  
  742. <------------>
  743. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Met hod: REGISTER)
  744.  
  745. <--- SIP read from UDP:162.197.74.233:53103 --->
  746. INVITE sip:13305344879@198.245.62.16:5060;transport=UDP SIP/2.0
  747. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---33c3c4d7bcf233ac ;rport
  748. Max-Forwards: 70
  749. Contact: <sip:12345424534@162.197.74.233:53103;transport=UDP>
  750. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>
  751. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  752. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  753. CSeq: 1 INVITE
  754. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  755. Content-Type: application/sdp
  756. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  757. User-Agent: Zoiper r30645
  758. Allow-Events: presence, kpml
  759. Content-Length: 245
  760.  
  761. v=0
  762. o=Zoiper 0 0 IN IP4 162.197.74.233
  763. s=Zoiper
  764. c=IN IP4 162.197.74.233
  765. t=0 0
  766. m=audio 60162 RTP/AVP 3 0 8 101
  767. a=rtpmap:3 GSM/8000
  768. a=rtpmap:0 PCMU/8000
  769. a=rtpmap:8 PCMA/8000
  770. a=rtpmap:101 telephone-event/8000
  771. a=fmtp:101 0-16
  772. a=sendrecv
  773. <------------->
  774. --- (14 headers 12 lines) ---
  775. Sending to 162.197.74.233:53103 (NAT)
  776. Sending to 162.197.74.233:53103 (NAT)
  777. Using INVITE request as basis request - -huQTbgbeCshXbR7-a0Ibw..
  778. Found peer '12345424534' for '12345424534' from 162.197.74.233:53103
  779.  
  780. <--- Reliably Transmitting (NAT) to 162.197.74.233:53103 --->
  781. SIP/2.0 401 Unauthorized
  782. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---33c3c4d7bcf233ac ;received=162.197.74.233;rport=53103
  783. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  784. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as0326c3ac
  785. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  786. CSeq: 1 INVITE
  787. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  788. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  789. Supported: replaces
  790. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61cf31b2"
  791. Content-Length: 0
  792.  
  793.  
  794. <------------>
  795. Scheduling destruction of SIP dialog '-huQTbgbeCshXbR7-a0Ibw..' in 32000 ms (Met hod: INVITE)
  796.  
  797. <--- SIP read from UDP:162.197.74.233:53103 --->
  798. ACK sip:13305344879@198.245.62.16:5060;transport=UDP SIP/2.0
  799. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---33c3c4d7bcf233ac ;rport
  800. Max-Forwards: 70
  801. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as0326c3ac
  802. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  803. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  804. CSeq: 1 ACK
  805. Content-Length: 0
  806.  
  807. <------------->
  808. --- (8 headers 0 lines) ---
  809.  
  810. <--- SIP read from UDP:162.197.74.233:53103 --->
  811. INVITE sip:13305344879@198.245.62.16:5060;transport=UDP SIP/2.0
  812. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;rport
  813. Max-Forwards: 70
  814. Contact: <sip:12345424534@162.197.74.233:53103;transport=UDP>
  815. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>
  816. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  817. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  818. CSeq: 2 INVITE
  819. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB E
  820. Content-Type: application/sdp
  821. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco- serviceuri
  822. User-Agent: Zoiper r30645
  823. Authorization: Digest username="12345424534",realm="asterisk",nonce="61cf31b2",u ri="sip:13305344879@198.245.62.16:5060;transport=UDP",response="1410d11317af24c5 d5cdb555e62cdef6",algorithm=MD5
  824. Allow-Events: presence, kpml
  825. Content-Length: 245
  826.  
  827. v=0
  828. o=Zoiper 0 0 IN IP4 162.197.74.233
  829. s=Zoiper
  830. c=IN IP4 162.197.74.233
  831. t=0 0
  832. m=audio 60162 RTP/AVP 3 0 8 101
  833. a=rtpmap:3 GSM/8000
  834. a=rtpmap:0 PCMU/8000
  835. a=rtpmap:8 PCMA/8000
  836. a=rtpmap:101 telephone-event/8000
  837. a=fmtp:101 0-16
  838. a=sendrecv
  839. <------------->
  840. --- (15 headers 12 lines) ---
  841. Sending to 162.197.74.233:53103 (NAT)
  842. Using INVITE request as basis request - -huQTbgbeCshXbR7-a0Ibw..
  843. Found peer '12345424534' for '12345424534' from 162.197.74.233:53103
  844. Found RTP audio format 3
  845. Found RTP audio format 0
  846. Found RTP audio format 8
  847. Found RTP audio format 101
  848. Found audio description format GSM for ID 3
  849. Found audio description format PCMU for ID 0
  850. Found audio description format PCMA for ID 8
  851. Found audio description format telephone-event for ID 101
  852. Capabilities: us - (ulaw|g729), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/tex t=(nothing), combined - (ulaw)
  853. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
  854. Peer audio RTP is at port 162.197.74.233:60162
  855. Looking for 13305344879 in outgoing (domain 198.245.62.16)
  856. list_route: hop: <sip:12345424534@162.197.74.233:53103;transport=UDP>
  857.  
  858. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  859. SIP/2.0 100 Trying
  860. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  861. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  862. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>
  863. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  864. CSeq: 2 INVITE
  865. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  866. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  867. Supported: replaces
  868. Contact: <sip:13305344879@127.0.0.1:5060>
  869. Content-Length: 0
  870.  
  871.  
  872. <------------>
  873. Audio is at 12460
  874. Adding codec 100003 (ulaw) to SDP
  875. Adding codec 100008 (g729) to SDP
  876. Adding non-codec 0x1 (telephone-event) to SDP
  877. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  878. INVITE sip:13305344879@sip.flowroute.com SIP/2.0
  879. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK20a15548;rport
  880. Max-Forwards: 70
  881. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  882. To: <sip:13305344879@sip.flowroute.com>
  883. Contact: <sip:12345424534@127.0.0.1:5060>
  884. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  885. CSeq: 102 INVITE
  886. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  887. Date: Tue, 28 Jul 2015 17:41:52 GMT
  888. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  889. Supported: replaces
  890. Content-Type: application/sdp
  891. Content-Length: 289
  892.  
  893. v=0
  894. o=root 880757134 880757134 IN IP4 127.0.0.1
  895. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  896. c=IN IP4 127.0.0.1
  897. t=0 0
  898. m=audio 12460 RTP/AVP 0 18 101
  899. a=rtpmap:0 PCMU/8000
  900. a=rtpmap:18 G729/8000
  901. a=fmtp:18 annexb=no
  902. a=rtpmap:101 telephone-event/8000
  903. a=fmtp:101 0-16
  904. a=ptime:20
  905. a=sendrecv
  906.  
  907. ---
  908.  
  909. <--- SIP read from UDP:216.115.69.144:5060 --->
  910. SIP/2.0 100 Trying
  911. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK20a15548;rport=5060;received=198.2 45.62.16
  912. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  913. To: <sip:13305344879@sip.flowroute.com>
  914. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  915. CSeq: 102 INVITE
  916. Content-Length: 0
  917.  
  918. <------------->
  919. --- (7 headers 0 lines) ---
  920.  
  921. <--- SIP read from UDP:216.115.69.144:5060 --->
  922. SIP/2.0 407 Proxy Authentication Required
  923. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK20a15548;rp ort=5060
  924. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  925. To: <sip:13305344879@sip.flowroute.com>;tag=a90f6ff38a5ea9aa0a4de2557847562b.642 6
  926. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  927. CSeq: 102 INVITE
  928. Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="VbfAEVW3vuWX4KSdwc3 c2MZPxYvl9yZj", qop="auth"
  929. Content-Length: 0
  930.  
  931. <------------->
  932. --- (8 headers 0 lines) ---
  933. Transmitting (NAT) to 216.115.69.144:5060:
  934. ACK sip:13305344879@sip.flowroute.com SIP/2.0
  935. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK20a15548;rport
  936. Max-Forwards: 70
  937. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  938. To: <sip:13305344879@sip.flowroute.com>;tag=a90f6ff38a5ea9aa0a4de2557847562b.642 6
  939. Contact: <sip:12345424534@127.0.0.1:5060>
  940. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  941. CSeq: 102 ACK
  942. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  943. Content-Length: 0
  944.  
  945.  
  946. ---
  947. Audio is at 12460
  948. Adding codec 100003 (ulaw) to SDP
  949. Adding codec 100008 (g729) to SDP
  950. Adding non-codec 0x1 (telephone-event) to SDP
  951. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  952. INVITE sip:13305344879@sip.flowroute.com SIP/2.0
  953. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7d5fed84;rport
  954. Max-Forwards: 70
  955. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  956. To: <sip:13305344879@sip.flowroute.com>
  957. Contact: <sip:12345424534@127.0.0.1:5060>
  958. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  959. CSeq: 103 INVITE
  960. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  961. Proxy-Authorization: Digest username="68511092", realm="sip.flowroute.com", algo rithm=MD5, uri="sip:13305344879@sip.flowroute.com", nonce="VbfAEVW3vuWX4KSdwc3c2 MZPxYvl9yZj", response="66441b209bbc714661570dd249c2457d", qop=auth, cnonce="79c 4e13f", nc=00000001
  962. Date: Tue, 28 Jul 2015 17:41:53 GMT
  963. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  964. Supported: replaces
  965. Content-Type: application/sdp
  966. Content-Length: 289
  967.  
  968. v=0
  969. o=root 880757134 880757135 IN IP4 127.0.0.1
  970. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  971. c=IN IP4 127.0.0.1
  972. t=0 0
  973. m=audio 12460 RTP/AVP 0 18 101
  974. a=rtpmap:0 PCMU/8000
  975. a=rtpmap:18 G729/8000
  976. a=fmtp:18 annexb=no
  977. a=rtpmap:101 telephone-event/8000
  978. a=fmtp:101 0-16
  979. a=ptime:20
  980. a=sendrecv
  981.  
  982. ---
  983.  
  984. <--- SIP read from UDP:216.115.69.144:5060 --->
  985. SIP/2.0 100 Trying
  986. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7d5fed84;rport=5060;received=198.2 45.62.16
  987. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  988. To: <sip:13305344879@sip.flowroute.com>
  989. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  990. CSeq: 103 INVITE
  991. Content-Length: 0
  992.  
  993. <------------->
  994. --- (7 headers 0 lines) ---
  995.  
  996. <--- SIP read from UDP:216.115.69.144:5060 --->
  997. SIP/2.0 183 Session Progress
  998. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  999. To: <sip:13305344879@sip.flowroute.com>;tag=SDh674599-233f250c-co4674-INS002
  1000. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK7d5fed84;rp ort=5060
  1001. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  1002. CSeq: 103 INVITE
  1003. Record-Route: <sip:216.115.69.133;lr>
  1004. Record-Route: <sip:216.115.69.144;lr>
  1005. Contact: <sip:+13305344879@65.98.237.158:5060;transport=udp>
  1006. Content-Type: application/sdp
  1007. Content-Length: 180
  1008.  
  1009. v=0
  1010. o=- 591340177 591340177 IN IP4 65.98.237.158
  1011. s=-
  1012. c=IN IP4 65.98.237.158
  1013. t=0 0
  1014. m=audio 10486 RTP/AVP 0 101
  1015. a=rtpmap:101 telephone-event/8000
  1016. a=fmtp:101 0-15
  1017. a=ptime:20
  1018. <------------->
  1019. --- (11 headers 9 lines) ---
  1020. list_route: hop: <sip:216.115.69.144;lr>
  1021. list_route: hop: <sip:216.115.69.133;lr>
  1022. Found RTP audio format 0
  1023. Found RTP audio format 101
  1024. Found audio description format telephone-event for ID 101
  1025. Capabilities: us - (ulaw|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothin g), combined - (ulaw)
  1026. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
  1027. Peer audio RTP is at port 65.98.237.158:10486
  1028. Audio is at 11842
  1029. Adding codec 100003 (ulaw) to SDP
  1030. Adding non-codec 0x1 (telephone-event) to SDP
  1031.  
  1032. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  1033. SIP/2.0 183 Session Progress
  1034. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1035. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1036. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1037. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1038. CSeq: 2 INVITE
  1039. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1040. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1041. Supported: replaces
  1042. Contact: <sip:13305344879@127.0.0.1:5060>
  1043. Content-Type: application/sdp
  1044. Content-Length: 244
  1045.  
  1046. v=0
  1047. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1048. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1049. c=IN IP4 127.0.0.1
  1050. t=0 0
  1051. m=audio 11842 RTP/AVP 0 101
  1052. a=rtpmap:0 PCMU/8000
  1053. a=rtpmap:101 telephone-event/8000
  1054. a=fmtp:101 0-16
  1055. a=ptime:20
  1056. a=sendrecv
  1057.  
  1058. <------------>
  1059. Retransmitting #9 (NAT) to 216.115.69.144:5060:
  1060. SIP/2.0 200 OK
  1061. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  1062. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  1063. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  1064. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  1065. Record-Route: <sip:216.115.69.144;lr>
  1066. Record-Route: <sip:216.115.69.132;lr>
  1067. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  1068. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  1069. Call-ID: 537699074_101142141@4.55.17.35
  1070. CSeq: 5324 INVITE
  1071. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1072. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1073. Supported: replaces
  1074. Contact: <sip:12345424534@127.0.0.1:5060>
  1075. Content-Type: application/sdp
  1076. Content-Length: 289
  1077.  
  1078. v=0
  1079. o=root 913076305 913076305 IN IP4 127.0.0.1
  1080. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1081. c=IN IP4 127.0.0.1
  1082. t=0 0
  1083. m=audio 17644 RTP/AVP 0 18 101
  1084. a=rtpmap:0 PCMU/8000
  1085. a=rtpmap:18 G729/8000
  1086. a=fmtp:18 annexb=no
  1087. a=rtpmap:101 telephone-event/8000
  1088. a=fmtp:101 0-16
  1089. a=ptime:20
  1090. a=sendrecv
  1091.  
  1092. ---
  1093. Retransmitting #10 (NAT) to 216.115.69.144:5060:
  1094. SIP/2.0 200 OK
  1095. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK64e5.21b8eca5a960d4a60902de1c91021 796.0;received=216.115.69.144;rport=5060
  1096. Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK64e5.847ae0a74d0d2361809097a0cb3e0 6b5.1
  1097. Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK64e5.443f2e8bfd82f1e469b73d6e43f31 e64.0
  1098. Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0cB65ef4d985828ba9d
  1099. Record-Route: <sip:216.115.69.144;lr>
  1100. Record-Route: <sip:216.115.69.132;lr>
  1101. From: "DUFFIELD DAVID " <sip:+13305344879@fl.gg>;tag=gK0c03fe76
  1102. To: <sip:+12345424534@fl.gg>;tag=as0d1df224
  1103. Call-ID: 537699074_101142141@4.55.17.35
  1104. CSeq: 5324 INVITE
  1105. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1107. Supported: replaces
  1108. Contact: <sip:12345424534@127.0.0.1:5060>
  1109. Content-Type: application/sdp
  1110. Content-Length: 289
  1111.  
  1112. v=0
  1113. o=root 913076305 913076305 IN IP4 127.0.0.1
  1114. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1115. c=IN IP4 127.0.0.1
  1116. t=0 0
  1117. m=audio 17644 RTP/AVP 0 18 101
  1118. a=rtpmap:0 PCMU/8000
  1119. a=rtpmap:18 G729/8000
  1120. a=fmtp:18 annexb=no
  1121. a=rtpmap:101 telephone-event/8000
  1122. a=fmtp:101 0-16
  1123. a=ptime:20
  1124. a=sendrecv
  1125.  
  1126. ---
  1127. [Jul 28 13:42:00] WARNING[23666]: chan_sip.c:4175 retrans_pkt: Retransmission ti meout reached on transmission 537699074_101142141@4.55.17.35 for seqno 5324 (Cri tical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmis sions
  1128. Packet timed out after 32000ms with no response
  1129. Really destroying SIP dialog '537699074_101142141@4.55.17.35' Method: INVITE
  1130.  
  1131. <--- SIP read from UDP:216.115.69.144:5060 --->
  1132. SIP/2.0 200 OK
  1133. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  1134. To: <sip:13305344879@sip.flowroute.com>;tag=SDh674599-233f250c-co4674-INS002
  1135. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK7d5fed84;rp ort=5060
  1136. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  1137. CSeq: 103 INVITE
  1138. Record-Route: <sip:216.115.69.133;lr>
  1139. Record-Route: <sip:216.115.69.144;lr>
  1140. Contact: <sip:+13305344879@65.98.237.158:5060;transport=udp>
  1141. Content-Type: application/sdp
  1142. Content-Length: 183
  1143.  
  1144. v=0
  1145. o=- 591340177 591340177 IN IP4 65.98.237.158
  1146. s=-
  1147. c=IN IP4 65.98.237.158
  1148. t=0 0
  1149. m=audio 10486 RTP/AVP 0 101
  1150. a=rtpmap:101 telephone-event/8000
  1151. a=fmtp:101 0-15
  1152. a=maxptime:20
  1153. <------------->
  1154. --- (11 headers 9 lines) ---
  1155. list_route: hop: <sip:216.115.69.144;lr>
  1156. list_route: hop: <sip:216.115.69.133;lr>
  1157. set_destination: Parsing <sip:216.115.69.144;lr> for address/port to send to
  1158. set_destination: set destination to 216.115.69.144:5060
  1159. Transmitting (NAT) to 216.115.69.144:5060:
  1160. ACK sip:+13305344879@65.98.237.158:5060;transport=udp SIP/2.0
  1161. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK67b6071f;rport
  1162. Route: <sip:216.115.69.144;lr>,<sip:216.115.69.133;lr>
  1163. Max-Forwards: 70
  1164. From: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  1165. To: <sip:13305344879@sip.flowroute.com>;tag=SDh674599-233f250c-co4674-INS002
  1166. Contact: <sip:12345424534@127.0.0.1:5060>
  1167. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  1168. CSeq: 103 ACK
  1169. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1170. Content-Length: 0
  1171.  
  1172.  
  1173. ---
  1174. Audio is at 11842
  1175. Adding codec 100003 (ulaw) to SDP
  1176. Adding non-codec 0x1 (telephone-event) to SDP
  1177.  
  1178. <--- Reliably Transmitting (NAT) to 162.197.74.233:53103 --->
  1179. SIP/2.0 200 OK
  1180. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1181. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1182. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1183. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1184. CSeq: 2 INVITE
  1185. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1186. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1187. Supported: replaces
  1188. Contact: <sip:13305344879@127.0.0.1:5060>
  1189. Content-Type: application/sdp
  1190. Content-Length: 244
  1191.  
  1192. v=0
  1193. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1194. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1195. c=IN IP4 127.0.0.1
  1196. t=0 0
  1197. m=audio 11842 RTP/AVP 0 101
  1198. a=rtpmap:0 PCMU/8000
  1199. a=rtpmap:101 telephone-event/8000
  1200. a=fmtp:101 0-16
  1201. a=ptime:20
  1202. a=sendrecv
  1203.  
  1204. <------------>
  1205. Retransmitting #1 (NAT) to 162.197.74.233:53103:
  1206. SIP/2.0 200 OK
  1207. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1208. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1209. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1210. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1211. CSeq: 2 INVITE
  1212. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1213. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1214. Supported: replaces
  1215. Contact: <sip:13305344879@127.0.0.1:5060>
  1216. Content-Type: application/sdp
  1217. Content-Length: 244
  1218.  
  1219. v=0
  1220. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1221. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1222. c=IN IP4 127.0.0.1
  1223. t=0 0
  1224. m=audio 11842 RTP/AVP 0 101
  1225. a=rtpmap:0 PCMU/8000
  1226. a=rtpmap:101 telephone-event/8000
  1227. a=fmtp:101 0-16
  1228. a=ptime:20
  1229. a=sendrecv
  1230.  
  1231. ---
  1232. Retransmitting #2 (NAT) to 162.197.74.233:53103:
  1233. SIP/2.0 200 OK
  1234. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1235. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1236. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1237. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1238. CSeq: 2 INVITE
  1239. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1240. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1241. Supported: replaces
  1242. Contact: <sip:13305344879@127.0.0.1:5060>
  1243. Content-Type: application/sdp
  1244. Content-Length: 244
  1245.  
  1246. v=0
  1247. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1248. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1249. c=IN IP4 127.0.0.1
  1250. t=0 0
  1251. m=audio 11842 RTP/AVP 0 101
  1252. a=rtpmap:0 PCMU/8000
  1253. a=rtpmap:101 telephone-event/8000
  1254. a=fmtp:101 0-16
  1255. a=ptime:20
  1256. a=sendrecv
  1257.  
  1258. ---
  1259. Retransmitting #3 (NAT) to 162.197.74.233:53103:
  1260. SIP/2.0 200 OK
  1261. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1262. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1263. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1264. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1265. CSeq: 2 INVITE
  1266. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1268. Supported: replaces
  1269. Contact: <sip:13305344879@127.0.0.1:5060>
  1270. Content-Type: application/sdp
  1271. Content-Length: 244
  1272.  
  1273. v=0
  1274. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1275. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1276. c=IN IP4 127.0.0.1
  1277. t=0 0
  1278. m=audio 11842 RTP/AVP 0 101
  1279. a=rtpmap:0 PCMU/8000
  1280. a=rtpmap:101 telephone-event/8000
  1281. a=fmtp:101 0-16
  1282. a=ptime:20
  1283. a=sendrecv
  1284.  
  1285. ---
  1286.  
  1287. <--- SIP read from UDP:216.115.69.144:5060 --->
  1288. BYE sip:2345424534@127.0.0.1:5060 SIP/2.0
  1289. Record-Route: <sip:216.115.69.144;lr>
  1290. Record-Route: <sip:216.115.69.133;lr>
  1291. From: <sip:13305344879@sip.flowroute.com>;tag=SDh674599-233f250c-co4674-INS002
  1292. To: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  1293. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKa2f5.31a4f31491ad3a98cf888c99d8a47 e20.0
  1294. Via: SIP/2.0/UDP 216.115.69.133;branch=z9hG4bKa2f5.bb9a5358ed039e872c16767a02bd8 3c3.0
  1295. Via: SIP/2.0/UDP 65.98.237.158:5060;branch=z9hG4bKkfb62p10582hfq4f64m0.1
  1296. Max-Forwards: 32
  1297. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  1298. CSeq: 467401 BYE
  1299. Content-Length: 0
  1300.  
  1301. <------------->
  1302. --- (12 headers 0 lines) ---
  1303. Sending to 216.115.69.144:5060 (NAT)
  1304. Scheduling destruction of SIP dialog '300ab4215c3ea5c5174d19026e4f4878@sip.flowr oute.com' in 32000 ms (Method: BYE)
  1305.  
  1306. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  1307. SIP/2.0 200 OK
  1308. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKa2f5.31a4f31491ad3a98cf888c99d8a47 e20.0;received=216.115.69.144;rport=5060
  1309. Via: SIP/2.0/UDP 216.115.69.133;branch=z9hG4bKa2f5.bb9a5358ed039e872c16767a02bd8 3c3.0
  1310. Via: SIP/2.0/UDP 65.98.237.158:5060;branch=z9hG4bKkfb62p10582hfq4f64m0.1
  1311. Record-Route: <sip:216.115.69.144;lr>
  1312. Record-Route: <sip:216.115.69.133;lr>
  1313. From: <sip:13305344879@sip.flowroute.com>;tag=SDh674599-233f250c-co4674-INS002
  1314. To: <sip:12345424534@sip.flowroute.com>;tag=as1a8408da
  1315. Call-ID: 300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com
  1316. CSeq: 467401 BYE
  1317. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1319. Supported: replaces
  1320. Content-Length: 0
  1321.  
  1322.  
  1323. <------------>
  1324. Scheduling destruction of SIP dialog '-huQTbgbeCshXbR7-a0Ibw..' in 32000 ms (Met hod: INVITE)
  1325.  
  1326. <--- SIP read from UDP:216.115.69.144:5060 --->
  1327. OPTIONS sip:198.245.62.16:5060 SIP/2.0
  1328. Max-Forwards: 10
  1329. Record-Route: <sip:216.115.69.144;lr>
  1330. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK299.12750b387feb5fb75e5cbd4a38397a c4.0
  1331. Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
  1332. Route: <sip:216.115.69.144;lr;received='sip:198.245.62.16:5060'>
  1333. From: sip:ping@invalid;tag=71145d59
  1334. To: sip:198.245.62.16:5060
  1335. Call-ID: 0c26e475-d7378168-0bb900a@70.167.153.136
  1336. CSeq: 1 OPTIONS
  1337. Content-Length: 0
  1338.  
  1339. <------------->
  1340. --- (11 headers 0 lines) ---
  1341. Sending to 216.115.69.144:5060 (NAT)
  1342. Looking for s in internal (domain 198.245.62.16)
  1343.  
  1344. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  1345. SIP/2.0 200 OK
  1346. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK299.12750b387feb5fb75e5cbd4a38397a c4.0;received=216.115.69.144;rport=5060
  1347. Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
  1348. Record-Route: <sip:216.115.69.144;lr>
  1349. From: sip:ping@invalid;tag=71145d59
  1350. To: sip:198.245.62.16:5060;tag=as64d4800a
  1351. Call-ID: 0c26e475-d7378168-0bb900a@70.167.153.136
  1352. CSeq: 1 OPTIONS
  1353. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1354. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1355. Supported: replaces
  1356. Contact: <sip:127.0.0.1:5060>
  1357. Accept: application/sdp
  1358. Content-Length: 0
  1359.  
  1360.  
  1361. <------------>
  1362. Scheduling destruction of SIP dialog '0c26e475-d7378168-0bb900a@70.167.153.136' in 32000 ms (Method: OPTIONS)
  1363. Retransmitting #4 (NAT) to 162.197.74.233:53103:
  1364. SIP/2.0 200 OK
  1365. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1366. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1367. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1368. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1369. CSeq: 2 INVITE
  1370. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1371. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1372. Supported: replaces
  1373. Contact: <sip:13305344879@127.0.0.1:5060>
  1374. Content-Type: application/sdp
  1375. Content-Length: 244
  1376.  
  1377. v=0
  1378. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1379. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1380. c=IN IP4 127.0.0.1
  1381. t=0 0
  1382. m=audio 11842 RTP/AVP 0 101
  1383. a=rtpmap:0 PCMU/8000
  1384. a=rtpmap:101 telephone-event/8000
  1385. a=fmtp:101 0-16
  1386. a=ptime:20
  1387. a=sendrecv
  1388.  
  1389. ---
  1390. Retransmitting #5 (NAT) to 162.197.74.233:53103:
  1391. SIP/2.0 200 OK
  1392. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1393. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1394. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1395. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1396. CSeq: 2 INVITE
  1397. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1399. Supported: replaces
  1400. Contact: <sip:13305344879@127.0.0.1:5060>
  1401. Content-Type: application/sdp
  1402. Content-Length: 244
  1403.  
  1404. v=0
  1405. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1406. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1407. c=IN IP4 127.0.0.1
  1408. t=0 0
  1409. m=audio 11842 RTP/AVP 0 101
  1410. a=rtpmap:0 PCMU/8000
  1411. a=rtpmap:101 telephone-event/8000
  1412. a=fmtp:101 0-16
  1413. a=ptime:20
  1414. a=sendrecv
  1415.  
  1416. ---
  1417. Retransmitting #6 (NAT) to 162.197.74.233:53103:
  1418. SIP/2.0 200 OK
  1419. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1420. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1421. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1422. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1423. CSeq: 2 INVITE
  1424. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1425. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1426. Supported: replaces
  1427. Contact: <sip:13305344879@127.0.0.1:5060>
  1428. Content-Type: application/sdp
  1429. Content-Length: 244
  1430.  
  1431. v=0
  1432. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1433. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1434. c=IN IP4 127.0.0.1
  1435. t=0 0
  1436. m=audio 11842 RTP/AVP 0 101
  1437. a=rtpmap:0 PCMU/8000
  1438. a=rtpmap:101 telephone-event/8000
  1439. a=fmtp:101 0-16
  1440. a=ptime:20
  1441. a=sendrecv
  1442.  
  1443. ---
  1444. Retransmitting #7 (NAT) to 162.197.74.233:53103:
  1445. SIP/2.0 200 OK
  1446. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1447. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1448. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1449. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1450. CSeq: 2 INVITE
  1451. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1452. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1453. Supported: replaces
  1454. Contact: <sip:13305344879@127.0.0.1:5060>
  1455. Content-Type: application/sdp
  1456. Content-Length: 244
  1457.  
  1458. v=0
  1459. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1460. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1461. c=IN IP4 127.0.0.1
  1462. t=0 0
  1463. m=audio 11842 RTP/AVP 0 101
  1464. a=rtpmap:0 PCMU/8000
  1465. a=rtpmap:101 telephone-event/8000
  1466. a=fmtp:101 0-16
  1467. a=ptime:20
  1468. a=sendrecv
  1469.  
  1470. ---
  1471. Really destroying SIP dialog '95f5da82-a91042f-3270852@216.115.69.131' Method: O PTIONS
  1472.  
  1473. <--- SIP read from UDP:162.197.74.233:53103 --->
  1474.  
  1475.  
  1476. <------------->
  1477. Really destroying SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' Method: REGISTER
  1478. Retransmitting #8 (NAT) to 162.197.74.233:53103:
  1479. SIP/2.0 200 OK
  1480. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1481. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1482. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1483. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1484. CSeq: 2 INVITE
  1485. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1486. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1487. Supported: replaces
  1488. Contact: <sip:13305344879@127.0.0.1:5060>
  1489. Content-Type: application/sdp
  1490. Content-Length: 244
  1491.  
  1492. v=0
  1493. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1494. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1495. c=IN IP4 127.0.0.1
  1496. t=0 0
  1497. m=audio 11842 RTP/AVP 0 101
  1498. a=rtpmap:0 PCMU/8000
  1499. a=rtpmap:101 telephone-event/8000
  1500. a=fmtp:101 0-16
  1501. a=ptime:20
  1502. a=sendrecv
  1503.  
  1504. ---
  1505. Retransmitting #9 (NAT) to 162.197.74.233:53103:
  1506. SIP/2.0 200 OK
  1507. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1508. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1509. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1510. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1511. CSeq: 2 INVITE
  1512. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1513. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1514. Supported: replaces
  1515. Contact: <sip:13305344879@127.0.0.1:5060>
  1516. Content-Type: application/sdp
  1517. Content-Length: 244
  1518.  
  1519. v=0
  1520. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1521. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1522. c=IN IP4 127.0.0.1
  1523. t=0 0
  1524. m=audio 11842 RTP/AVP 0 101
  1525. a=rtpmap:0 PCMU/8000
  1526. a=rtpmap:101 telephone-event/8000
  1527. a=fmtp:101 0-16
  1528. a=ptime:20
  1529. a=sendrecv
  1530.  
  1531. ---
  1532. Retransmitting #10 (NAT) to 162.197.74.233:53103:
  1533. SIP/2.0 200 OK
  1534. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---469093b5716d7095 ;received=162.197.74.233;rport=53103
  1535. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=8452b77b
  1536. To: <sip:13305344879@198.245.62.16:5060;transport=UDP>;tag=as333a51c5
  1537. Call-ID: -huQTbgbeCshXbR7-a0Ibw..
  1538. CSeq: 2 INVITE
  1539. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1540. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
  1541. Supported: replaces
  1542. Contact: <sip:13305344879@127.0.0.1:5060>
  1543. Content-Type: application/sdp
  1544. Content-Length: 244
  1545.  
  1546. v=0
  1547. o=root 1185474627 1185474627 IN IP4 127.0.0.1
  1548. s=Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1549. c=IN IP4 127.0.0.1
  1550. t=0 0
  1551. m=audio 11842 RTP/AVP 0 101
  1552. a=rtpmap:0 PCMU/8000
  1553. a=rtpmap:101 telephone-event/8000
  1554. a=fmtp:101 0-16
  1555. a=ptime:20
  1556. a=sendrecv
  1557.  
  1558. ---
  1559. [Jul 28 13:42:33] WARNING[23666]: chan_sip.c:4175 retrans_pkt: Retransmission ti meout reached on transmission -huQTbgbeCshXbR7-a0Ibw.. for seqno 2 (Critical Res ponse) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  1560. Packet timed out after 32000ms with no response
  1561. Really destroying SIP dialog '-huQTbgbeCshXbR7-a0Ibw..' Method: INVITE
  1562. Really destroying SIP dialog '300ab4215c3ea5c5174d19026e4f4878@sip.flowroute.com' Method: BYE
  1563. Really destroying SIP dialog '0c26e475-d7378168-0bb900a@70.167.153.136' Method: OPTIONS
  1564.  
  1565. <--- SIP read from UDP:216.115.69.144:5060 --->
  1566. OPTIONS sip:198.245.62.16:5060 SIP/2.0
  1567. Max-Forwards: 10
  1568. Record-Route: <sip:216.115.69.144;lr>
  1569. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK320c.4fef39eb065eb1b7234d99a8c8b5f3fc.0
  1570. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  1571. Route: <sip:216.115.69.144;lr;received='sip:198.245.62.16:5060'>
  1572. From: sip:ping@invalid;tag=e1d1e61f
  1573. To: sip:198.245.62.16:5060
  1574. Call-ID: 95f5da82-465242f-a570852@216.115.69.131
  1575. CSeq: 1 OPTIONS
  1576. Content-Length: 0
  1577.  
  1578. <------------->
  1579. --- (11 headers 0 lines) ---
  1580. Sending to 216.115.69.144:5060 (NAT)
  1581. Looking for s in internal (domain 198.245.62.16)
  1582.  
  1583. <--- Transmitting (NAT) to 216.115.69.144:5060 --->
  1584. SIP/2.0 200 OK
  1585. Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK320c.4fef39eb065eb1b7234d99a8c8b5f3fc.0;received=216.115.69.144;rport=5060
  1586. Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
  1587. Record-Route: <sip:216.115.69.144;lr>
  1588. From: sip:ping@invalid;tag=e1d1e61f
  1589. To: sip:198.245.62.16:5060;tag=as372594b4
  1590. Call-ID: 95f5da82-465242f-a570852@216.115.69.131
  1591. CSeq: 1 OPTIONS
  1592. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1593. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1594. Supported: replaces
  1595. Contact: <sip:127.0.0.1:5060>
  1596. Accept: application/sdp
  1597. Content-Length: 0
  1598.  
  1599.  
  1600. <------------>
  1601. Scheduling destruction of SIP dialog '95f5da82-465242f-a570852@216.115.69.131' in 32000 ms (Method: OPTIONS)
  1602.  
  1603. <--- SIP read from UDP:162.197.74.233:53103 --->
  1604. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  1605. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---2a62b5bab0a71158;rport
  1606. Max-Forwards: 70
  1607. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transport=UDP>
  1608. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  1609. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  1610. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  1611. CSeq: 5 REGISTER
  1612. Expires: 60
  1613. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  1614. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
  1615. User-Agent: Zoiper r30645
  1616. Authorization: Digest username="12345424534",realm="asterisk",nonce="3852a490",uri="sip:198.245.62.16:5060;transport=UDP",response="c99faeaf0c4c2938f01674aff06f8520",algorithm=MD5
  1617. Allow-Events: presence, kpml
  1618. Content-Length: 0
  1619.  
  1620. <------------->
  1621. --- (15 headers 0 lines) ---
  1622. Sending to 162.197.74.233:53103 (NAT)
  1623. Sending to 162.197.74.233:53103 (NAT)
  1624.  
  1625. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  1626. SIP/2.0 401 Unauthorized
  1627. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---2a62b5bab0a71158;received=162.197.74.233;rport=53103
  1628. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  1629. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as4a94f887
  1630. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  1631. CSeq: 5 REGISTER
  1632. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1633. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1634. Supported: replaces
  1635. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="254407e7"
  1636. Content-Length: 0
  1637.  
  1638.  
  1639. <------------>
  1640. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Method: REGISTER)
  1641.  
  1642. <--- SIP read from UDP:162.197.74.233:53103 --->
  1643. REGISTER sip:198.245.62.16:5060;transport=UDP SIP/2.0
  1644. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---2d14564dd234800b;rport
  1645. Max-Forwards: 70
  1646. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transport=UDP>
  1647. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>
  1648. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  1649. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  1650. CSeq: 6 REGISTER
  1651. Expires: 60
  1652. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  1653. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
  1654. User-Agent: Zoiper r30645
  1655. Authorization: Digest username="12345424534",realm="asterisk",nonce="254407e7",uri="sip:198.245.62.16:5060;transport=UDP",response="1733f92ff98fc8808d4bb5a8e2b3dc14",algorithm=MD5
  1656. Allow-Events: presence, kpml
  1657. Content-Length: 0
  1658.  
  1659. <------------->
  1660. --- (15 headers 0 lines) ---
  1661. Sending to 162.197.74.233:53103 (NAT)
  1662.  
  1663. <--- Transmitting (NAT) to 162.197.74.233:53103 --->
  1664. SIP/2.0 200 OK
  1665. Via: SIP/2.0/UDP 162.197.74.233:53103;branch=z9hG4bK-524287-1---2d14564dd234800b;received=162.197.74.233;rport=53103
  1666. From: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=efa3c417
  1667. To: <sip:12345424534@198.245.62.16:5060;transport=UDP>;tag=as4a94f887
  1668. Call-ID: QqRkOzCLAuzdi3JmwAbJ7w..
  1669. CSeq: 6 REGISTER
  1670. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1671. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1672. Supported: replaces
  1673. Expires: 60
  1674. Contact: <sip:12345424534@162.197.74.233:53103;rinstance=609662a68d43d511;transport=UDP>;expires=60
  1675. Date: Tue, 28 Jul 2015 17:42:48 GMT
  1676. Content-Length: 0
  1677.  
  1678.  
  1679. <------------>
  1680. Scheduling destruction of SIP dialog 'QqRkOzCLAuzdi3JmwAbJ7w..' in 32000 ms (Method: REGISTER)
  1681.  
  1682. <--- SIP read from UDP:23.92.94.65:5074 --->
  1683. INVITE sip:9011972595452091@198.245.62.16 SIP/2.0
  1684. To: 9011972595452091<sip:9011972595452091@198.245.62.16>
  1685. From: 51001<sip:51001@198.245.62.16>;tag=4f7910ea
  1686. Via: SIP/2.0/UDP 23.92.94.65:5074;branch=z9hG4bK-ed2bb1452612fb790c8e3d51aa2f29fa;rport
  1687. Call-ID: ed2bb1452612fb790c8e3d51aa2f29fa
  1688. CSeq: 1 INVITE
  1689. Contact: <sip:51001@23.92.94.65:5074>
  1690. Max-Forwards: 70
  1691. Allow: INVITE, ACK, CANCEL, BYE
  1692. User-Agent: sipcli/v1.8
  1693. Content-Type: application/sdp
  1694. Content-Length: 278
  1695.  
  1696. v=0
  1697. o=sipcli-Session 1406391765 418372928 IN IP4 23.92.94.65
  1698. s=sipcli
  1699. c=IN IP4 23.92.94.65
  1700. t=0 0
  1701. m=audio 5075 RTP/AVP 18 0 8 101
  1702. a=fmtp:101 0-15
  1703. a=rtpmap:18 G729/8000
  1704. a=rtpmap:0 PCMU/8000
  1705. a=rtpmap:8 PCMA/8000
  1706. a=rtpmap:101 telephone-event/8000
  1707. a=ptime:20
  1708. a=sendrecv
  1709. <------------->
  1710. --- (12 headers 13 lines) ---
  1711. Sending to 23.92.94.65:5074 (NAT)
  1712. Sending to 23.92.94.65:5074 (NAT)
  1713. Using INVITE request as basis request - ed2bb1452612fb790c8e3d51aa2f29fa
  1714. No matching peer for '51001' from '23.92.94.65:5074'
  1715.  
  1716. <--- Reliably Transmitting (NAT) to 23.92.94.65:5074 --->
  1717. SIP/2.0 401 Unauthorized
  1718. Via: SIP/2.0/UDP 23.92.94.65:5074;branch=z9hG4bK-ed2bb1452612fb790c8e3d51aa2f29fa;received=23.92.94.65;rport=5074
  1719. From: 51001<sip:51001@198.245.62.16>;tag=4f7910ea
  1720. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as52e3c5de
  1721. Call-ID: ed2bb1452612fb790c8e3d51aa2f29fa
  1722. CSeq: 1 INVITE
  1723. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1724. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1725. Supported: replaces
  1726. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="533abffe"
  1727. Content-Length: 0
  1728.  
  1729.  
  1730. <------------>
  1731. Scheduling destruction of SIP dialog 'ed2bb1452612fb790c8e3d51aa2f29fa' in 32000 ms (Method: INVITE)
  1732. Retransmitting #1 (NAT) to 23.92.94.65:5074:
  1733. SIP/2.0 401 Unauthorized
  1734. Via: SIP/2.0/UDP 23.92.94.65:5074;branch=z9hG4bK-ed2bb1452612fb790c8e3d51aa2f29fa;received=23.92.94.65;rport=5074
  1735. From: 51001<sip:51001@198.245.62.16>;tag=4f7910ea
  1736. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as52e3c5de
  1737. Call-ID: ed2bb1452612fb790c8e3d51aa2f29fa
  1738. CSeq: 1 INVITE
  1739. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1740. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1741. Supported: replaces
  1742. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="533abffe"
  1743. Content-Length: 0
  1744.  
  1745.  
  1746. ---
  1747. Retransmitting #2 (NAT) to 23.92.94.65:5074:
  1748. SIP/2.0 401 Unauthorized
  1749. Via: SIP/2.0/UDP 23.92.94.65:5074;branch=z9hG4bK-ed2bb1452612fb790c8e3d51aa2f29fa;received=23.92.94.65;rport=5074
  1750. From: 51001<sip:51001@198.245.62.16>;tag=4f7910ea
  1751. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as52e3c5de
  1752. Call-ID: ed2bb1452612fb790c8e3d51aa2f29fa
  1753. CSeq: 1 INVITE
  1754. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1755. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1756. Supported: replaces
  1757. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="533abffe"
  1758. Content-Length: 0
  1759.  
  1760.  
  1761. ---
  1762. [Jul 28 13:42:54] NOTICE[23666]: chan_sip.c:15142 sip_reregister: -- Re-registration for 68511092@sip.flowroute.com
  1763. REGISTER 11 headers, 0 lines
  1764. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  1765. REGISTER sip:sip.flowroute.com SIP/2.0
  1766. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK32532e7b;rport
  1767. Max-Forwards: 70
  1768. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  1769. To: <sip:68511092@sip.flowroute.com>
  1770. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  1771. CSeq: 654 REGISTER
  1772. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1773. Authorization: Digest username="68511092", realm="sip.flowroute.com", algorithm=MD5, uri="sip:sip.flowroute.com", nonce="Vbe/FFW3veiMzl1QiinyvfKFw+jIhuGh", response="3653c9eecd07eb499b82211ade0247ea", qop=auth, cnonce="06eb2d21", nc=00000004
  1774. Expires: 120
  1775. Contact: <sip:s@127.0.0.1:5060>
  1776. Content-Length: 0
  1777.  
  1778.  
  1779. ---
  1780.  
  1781. <--- SIP read from UDP:216.115.69.144:5060 --->
  1782. SIP/2.0 401 Unauthorized
  1783. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK32532e7b;rport=5060
  1784. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  1785. To: <sip:68511092@sip.flowroute.com>;tag=aa681f9fdf30149b00040f579a1d99c4.f209
  1786. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  1787. CSeq: 654 REGISTER
  1788. WWW-Authenticate: Digest realm="sip.flowroute.com", nonce="VbfAT1W3vyMULFPRuPWprHBRr+C6/LE5", qop="auth"
  1789. Content-Length: 0
  1790.  
  1791. <------------->
  1792. --- (8 headers 0 lines) ---
  1793. Responding to challenge, registration to domain/host name sip.flowroute.com
  1794. REGISTER 11 headers, 0 lines
  1795. Reliably Transmitting (NAT) to 216.115.69.144:5060:
  1796. REGISTER sip:sip.flowroute.com SIP/2.0
  1797. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3f30ea64;rport
  1798. Max-Forwards: 70
  1799. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  1800. To: <sip:68511092@sip.flowroute.com>
  1801. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  1802. CSeq: 655 REGISTER
  1803. User-Agent: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1804. Authorization: Digest username="68511092", realm="sip.flowroute.com", algorithm=MD5, uri="sip:sip.flowroute.com", nonce="VbfAT1W3vyMULFPRuPWprHBRr+C6/LE5", response="59c79a44e04ef811b016758c813bf77f", qop=auth, cnonce="00747278", nc=00000001
  1805. Expires: 120
  1806. Contact: <sip:s@127.0.0.1:5060>
  1807. Content-Length: 0
  1808.  
  1809.  
  1810. ---
  1811.  
  1812. <--- SIP read from UDP:216.115.69.144:5060 --->
  1813. SIP/2.0 200 OK
  1814. Via: SIP/2.0/UDP 127.0.0.1:5060;received=198.245.62.16;branch=z9hG4bK3f30ea64;rport=5060
  1815. From: <sip:68511092@sip.flowroute.com>;tag=as1a44f4a9
  1816. To: <sip:68511092@sip.flowroute.com>;tag=aa681f9fdf30149b00040f579a1d99c4.34a1
  1817. Call-ID: 02bb0a1d18d326ac35294d9425595a75@[::1]
  1818. CSeq: 655 REGISTER
  1819. Contact: <sip:s@127.0.0.1:5060>;q=1;expires=120;received="sip:198.245.62.16:5060"
  1820. Content-Length: 0
  1821.  
  1822. <------------->
  1823. --- (8 headers 0 lines) ---
  1824. [Jul 28 13:42:54] NOTICE[23666]: chan_sip.c:23655 handle_response_register: Outbound Registration: Expiry for sip.flowroute.com is 120 sec (Scheduling reregistration in 105 s)
  1825. Really destroying SIP dialog '02bb0a1d18d326ac35294d9425595a75@[::1]' Method: REGISTER
  1826. Retransmitting #3 (NAT) to 23.92.94.65:5074:
  1827. SIP/2.0 401 Unauthorized
  1828. Via: SIP/2.0/UDP 23.92.94.65:5074;branch=z9hG4bK-ed2bb1452612fb790c8e3d51aa2f29fa;received=23.92.94.65;rport=5074
  1829. From: 51001<sip:51001@198.245.62.16>;tag=4f7910ea
  1830. To: 9011972595452091<sip:9011972595452091@198.245.62.16>;tag=as52e3c5de
  1831. Call-ID: ed2bb1452612fb790c8e3d51aa2f29fa
  1832. CSeq: 1 INVITE
  1833. Server: Asterisk PBX 11.11.0~dfsg-2ubuntu1
  1834. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1835. Supported: replaces
  1836. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="533abffe"
  1837. Content-Length: 0
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