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  1. <------------->
  2. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c: --- (7 headers 0 lines) ---
  3. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c:
  4. <--- SIP read from UDP:192.168.10.247:5060 --->
  5. INVITE sip:126@255.255.255.0 SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK80fc00e45a89e5118c72e6d1e637b86c;rport
  7. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  8. To: <sip:126@255.255.255.0>
  9. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  10. CSeq: 10 INVITE
  11. Contact: <sip:125@192.168.10.247:5060>
  12. Authorization: Digest username="125", realm="asterisk", nonce="57f8de14", uri="sip:126@255.255.255.0", response="2879ed41f9a6ec185837a861077c638d", algorithm=MD5
  13. Content-Type: application/sdp
  14. Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE
  15. Max-Forwards: 70
  16. Supported: 100rel, replaces, from-change
  17. P-Early-Media: supported
  18. User-Agent: SIPPER for PhonerLite
  19. P-Preferred-Identity: <sip:125@255.255.255.0>
  20. Content-Length: 257
  21.  
  22. v=0
  23. o=- 1661247578 1 IN IP4 192.168.10.247
  24. s=SIPPER for PhonerLite
  25. c=IN IP4 192.168.10.247
  26. t=0 0
  27. m=audio 5062 RTP/AVP 8 0 101
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:0 PCMU/8000
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-16
  32. a=ssrc:2071226455
  33. a=sendrecv
  34. <------------->
  35. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c: --- (16 headers 12 lines) ---
  36. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Sending to 192.168.10.247:5060 (NAT)
  37. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Using INVITE request as basis request - 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  38. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found peer '125' for '125' from 192.168.10.247:5060
  39. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found RTP audio format 8
  40. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found RTP audio format 0
  41. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found RTP audio format 101
  42. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found audio description format PCMA for ID 8
  43. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found audio description format PCMU for ID 0
  44. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Found audio description format telephone-event for ID 101
  45. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  46. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  47. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Peer audio RTP is at port 192.168.10.247:5062
  48. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: Looking for 126 in from-internal (domain 255.255.255.0)
  49. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: list_route: hop: <sip:125@192.168.10.247:5060>
  50. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c:
  51. <--- Transmitting (NAT) to 192.168.10.247:5060 --->
  52. SIP/2.0 100 Trying
  53. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK80fc00e45a89e5118c72e6d1e637b86c;received=192.168.10.247;rport=5060
  54. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  55. To: <sip:126@255.255.255.0>
  56. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  57. CSeq: 10 INVITE
  58. Server: FPBX-2.11.0(11.17.1)
  59. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  60. Supported: replaces, timer
  61. Contact: <sip:126@192.168.10.225:5060>
  62. Content-Length: 0
  63.  
  64.  
  65. <------------>
  66. [2015-11-14 19:31:54] WARNING[17207][C-0000029d] func_presencestate.c: PRESENCE_STATE unknown
  67. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Audio is at 18846
  68. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Adding codec 100004 (alaw) to SDP
  69. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Adding codec 100002 (gsm) to SDP
  70. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Adding codec 100003 (ulaw) to SDP
  71. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  72. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.247:53140:
  73. INVITE sip:126@192.168.10.247:53140;ob SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5b7adfa0;rport
  75. Max-Forwards: 70
  76. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  77. To: <sip:126@192.168.10.247:53140;ob>
  78. Contact: <sip:125@192.168.10.225:5060>
  79. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  80. CSeq: 102 INVITE
  81. User-Agent: FPBX-2.11.0(11.17.1)
  82. Date: Sat, 14 Nov 2015 16:31:54 GMT
  83. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  84. Supported: replaces, timer
  85. Content-Type: application/sdp
  86. Content-Length: 285
  87.  
  88. v=0
  89. o=root 1202606172 1202606172 IN IP4 192.168.10.225
  90. s=Asterisk PBX 11.17.1
  91. c=IN IP4 192.168.10.225
  92. t=0 0
  93. m=audio 18846 RTP/AVP 8 3 0 101
  94. a=rtpmap:8 PCMA/8000
  95. a=rtpmap:3 GSM/8000
  96. a=rtpmap:0 PCMU/8000
  97. a=rtpmap:101 telephone-event/8000
  98. a=fmtp:101 0-16
  99. a=ptime:20
  100. a=sendrecv
  101.  
  102. ---
  103. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c:
  104. <--- Transmitting (NAT) to 192.168.10.247:5060 --->
  105. SIP/2.0 180 Ringing
  106. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK80fc00e45a89e5118c72e6d1e637b86c;received=192.168.10.247;rport=5060
  107. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  108. To: <sip:126@255.255.255.0>;tag=as03439ec7
  109. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  110. CSeq: 10 INVITE
  111. Server: FPBX-2.11.0(11.17.1)
  112. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  113. Supported: replaces, timer
  114. Contact: <sip:126@192.168.10.225:5060>
  115. Content-Length: 0
  116.  
  117.  
  118. <------------>
  119. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c: Retransmitting #1 (NAT) to 192.168.10.247:53140:
  120. INVITE sip:126@192.168.10.247:53140;ob SIP/2.0
  121. Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5b7adfa0;rport
  122. Max-Forwards: 70
  123. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  124. To: <sip:126@192.168.10.247:53140;ob>
  125. Contact: <sip:125@192.168.10.225:5060>
  126. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  127. CSeq: 102 INVITE
  128. User-Agent: FPBX-2.11.0(11.17.1)
  129. Date: Sat, 14 Nov 2015 16:31:54 GMT
  130. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  131. Supported: replaces, timer
  132. Content-Type: application/sdp
  133. Content-Length: 285
  134.  
  135. v=0
  136. o=root 1202606172 1202606172 IN IP4 192.168.10.225
  137. s=Asterisk PBX 11.17.1
  138. c=IN IP4 192.168.10.225
  139. t=0 0
  140. m=audio 18846 RTP/AVP 8 3 0 101
  141. a=rtpmap:8 PCMA/8000
  142. a=rtpmap:3 GSM/8000
  143. a=rtpmap:0 PCMU/8000
  144. a=rtpmap:101 telephone-event/8000
  145. a=fmtp:101 0-16
  146. a=ptime:20
  147. a=sendrecv
  148.  
  149. ---
  150. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c:
  151. <--- SIP read from UDP:192.168.10.247:53140 --->
  152. SIP/2.0 100 Trying
  153. Via: SIP/2.0/UDP 192.168.10.225:5060;rport=5060;received=192.168.10.225;branch=z9hG4bK5b7adfa0
  154. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  155. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  156. To: <sip:126@192.168.10.247;ob>
  157. CSeq: 102 INVITE
  158. Content-Length: 0
  159.  
  160. <------------->
  161. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c: --- (7 headers 0 lines) ---
  162. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c:
  163. <--- SIP read from UDP:192.168.10.247:53140 --->
  164. SIP/2.0 180 Ringing
  165. Via: SIP/2.0/UDP 192.168.10.225:5060;rport=5060;received=192.168.10.225;branch=z9hG4bK5b7adfa0
  166. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  167. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  168. To: <sip:126@192.168.10.247;ob>;tag=5b8f504a84be4ac3a5ed9582645f2029
  169. CSeq: 102 INVITE
  170. Contact: <sip:192.168.10.247:53140>
  171. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  172. Content-Length: 0
  173.  
  174. <------------->
  175. [2015-11-14 19:31:54] VERBOSE[3079] chan_sip.c: --- (9 headers 0 lines) ---
  176. [2015-11-14 19:31:54] VERBOSE[3079][C-0000029d] chan_sip.c: list_route: hop: <sip:192.168.10.247:53140>
  177. [2015-11-14 19:31:54] VERBOSE[17207][C-0000029d] chan_sip.c:
  178. <--- Transmitting (NAT) to 192.168.10.247:5060 --->
  179. SIP/2.0 180 Ringing
  180. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK80fc00e45a89e5118c72e6d1e637b86c;received=192.168.10.247;rport=5060
  181. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  182. To: <sip:126@255.255.255.0>;tag=as03439ec7
  183. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  184. CSeq: 10 INVITE
  185. Server: FPBX-2.11.0(11.17.1)
  186. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  187. Supported: replaces, timer
  188. Contact: <sip:126@192.168.10.225:5060>
  189. Content-Length: 0
  190.  
  191.  
  192. <------------>
  193. [2015-11-14 19:31:57] VERBOSE[3079] chan_sip.c:
  194. <--- SIP read from UDP:192.168.10.247:53140 --->
  195. SIP/2.0 200 OK
  196. Via: SIP/2.0/UDP 192.168.10.225:5060;rport=5060;received=192.168.10.225;branch=z9hG4bK5b7adfa0
  197. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  198. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  199. To: <sip:126@192.168.10.247;ob>;tag=5b8f504a84be4ac3a5ed9582645f2029
  200. CSeq: 102 INVITE
  201. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  202. Contact: <sip:192.168.10.247:53140>
  203. Supported: replaces, 100rel, timer, norefersub
  204. Content-Type: application/sdp
  205. Content-Length: 277
  206.  
  207. v=0
  208. o=- 3656518317 3656518318 IN IP4 192.168.10.247
  209. s=pjmedia
  210. b=AS:84
  211. t=0 0
  212. a=X-nat:0
  213. m=audio 4006 RTP/AVP 8 101
  214. c=IN IP4 192.168.10.247
  215. b=TIAS:64000
  216. a=rtcp:4007 IN IP4 192.168.10.247
  217. a=sendrecv
  218. a=rtpmap:8 PCMA/8000
  219. a=rtpmap:101 telephone-event/8000
  220. a=fmtp:101 0-16
  221. <------------->
  222. [2015-11-14 19:31:57] VERBOSE[3079] chan_sip.c: --- (11 headers 14 lines) ---
  223. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Found RTP audio format 8
  224. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Found RTP audio format 101
  225. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Found audio description format PCMA for ID 8
  226. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Found audio description format telephone-event for ID 101
  227. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  228. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  229. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Peer audio RTP is at port 192.168.10.247:4006
  230. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: list_route: hop: <sip:192.168.10.247:53140>
  231. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: set_destination: Parsing <sip:192.168.10.247:53140> for address/port to send to
  232. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: set_destination: set destination to 192.168.10.247:53140
  233. [2015-11-14 19:31:57] VERBOSE[3079][C-0000029d] chan_sip.c: Transmitting (NAT) to 192.168.10.247:53140:
  234. ACK sip:192.168.10.247:53140 SIP/2.0
  235. Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2c9157e4;rport
  236. Max-Forwards: 70
  237. From: "125" <sip:125@192.168.10.225>;tag=as06ea4bfe
  238. To: <sip:126@192.168.10.247:53140;ob>;tag=5b8f504a84be4ac3a5ed9582645f2029
  239. Contact: <sip:125@192.168.10.225:5060>
  240. Call-ID: 643431670cd81c91573783cf2e9cad99@192.168.10.225:5060
  241. CSeq: 102 ACK
  242. User-Agent: FPBX-2.11.0(11.17.1)
  243. Content-Length: 0
  244.  
  245.  
  246. ---
  247. [2015-11-14 19:31:57] VERBOSE[17207][C-0000029d] chan_sip.c: Audio is at 16200
  248. [2015-11-14 19:31:57] VERBOSE[17207][C-0000029d] chan_sip.c: Adding codec 100004 (alaw) to SDP
  249. [2015-11-14 19:31:57] VERBOSE[17207][C-0000029d] chan_sip.c: Adding codec 100003 (ulaw) to SDP
  250. [2015-11-14 19:31:57] VERBOSE[17207][C-0000029d] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  251. [2015-11-14 19:31:57] VERBOSE[17207][C-0000029d] chan_sip.c:
  252. <--- Reliably Transmitting (NAT) to 192.168.10.247:5060 --->
  253. SIP/2.0 200 OK
  254. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK80fc00e45a89e5118c72e6d1e637b86c;received=192.168.10.247;rport=5060
  255. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  256. To: <sip:126@255.255.255.0>;tag=as03439ec7
  257. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  258. CSeq: 10 INVITE
  259. Server: FPBX-2.11.0(11.17.1)
  260. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  261. Supported: replaces, timer
  262. Contact: <sip:126@192.168.10.225:5060>
  263. Content-Type: application/sdp
  264. Content-Length: 260
  265.  
  266. v=0
  267. o=root 310746018 310746018 IN IP4 192.168.10.225
  268. s=Asterisk PBX 11.17.1
  269. c=IN IP4 192.168.10.225
  270. t=0 0
  271. m=audio 16200 RTP/AVP 8 0 101
  272. a=rtpmap:8 PCMA/8000
  273. a=rtpmap:0 PCMU/8000
  274. a=rtpmap:101 telephone-event/8000
  275. a=fmtp:101 0-16
  276. a=ptime:20
  277. a=sendrecv
  278.  
  279. <------------>
  280. [2015-11-14 19:31:57] VERBOSE[3079] chan_sip.c:
  281. <--- SIP read from UDP:192.168.10.247:5060 --->
  282. ACK sip:126@192.168.10.225:5060 SIP/2.0
  283. Via: SIP/2.0/UDP 192.168.10.247:5060;branch=z9hG4bK802932e55a89e5118c72e6d1e637b86c;rport
  284. From: "125" <sip:125@255.255.255.0>;tag=1058336498
  285. To: <sip:126@255.255.255.0>;tag=as03439ec7
  286. Call-ID: 80FC00E4-5A89-E511-8C70-E6D1E637B86C@192.168.10.247
  287. CSeq: 10 ACK
  288. Contact: <sip:125@192.168.10.247:5060>
  289. Authorization: Digest username="125", realm="asterisk", nonce="57f8de14", uri="sip:126@255.255.255.0", response="2879ed41f9a6ec185837a861077c638d", algorithm=MD5
  290. Max-Forwards: 70
  291. Content-Length: 0
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