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  1. root@Asterisk-Server:/etc/asterisk# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
  2. Asterisk 13.6.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  3. Created by Mark Spencer <[email protected]>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Running as user 'asteriskpbx'
  10. Running under group 'asteriskpbx'
  11. Connected to Asterisk 13.6.0 currently running on Asterisk-Server (pid = 8321)
  12.  
  13. <--- SIP read from UDP:192.168.1.102:5070 --->
  14. INVITE sip:[email protected] SIP/2.0
  15. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1656108
  16. To: <sip:[email protected]>
  17. From: "Stephen" <sip:[email protected]>;tag=8325
  18. Call-ID: [email protected]
  19. CSeq: 329 INVITE
  20. Max-Forwards: 70
  21. User-Agent: NCH Software Express Talk 4.35
  22. Contact: <sip:[email protected]:5070>
  23. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  24. Supported: replaces
  25. Content-Type: application/sdp
  26. Content-Length: 354
  27.  
  28. v=0
  29. o=NCHSoftware-Talk 1447514436 1447514449 IN IP4 192.168.1.102
  30. s=Express Talk Call
  31. c=IN IP4 192.168.1.102
  32. t=0 0
  33. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  34. a=rtpmap:0 PCMU/8000
  35. a=rtpmap:8 PCMA/8000
  36. a=rtpmap:96 G726-32/8000
  37. a=rtpmap:3 GSM/8000
  38. a=rtpmap:13 CN/8000
  39. a=rtpmap:101 telephone-event/8000
  40. a=fmtp:101 0-16
  41. a=sendrecv
  42. a=direction:active
  43. <------------->
  44. --- (13 headers 15 lines) ---
  45. Sending to 192.168.1.102:5070 (no NAT)
  46. Sending to 192.168.1.102:5070 (no NAT)
  47. Using INVITE request as basis request - [email protected]
  48. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  49.  
  50. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  51. SIP/2.0 401 Unauthorized
  52. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1656108;received=192.168.1.102;rport=5070
  53. From: "Stephen" <sip:[email protected]>;tag=8325
  54. To: <sip:[email protected]>;tag=as253f2ec7
  55. Call-ID: [email protected]
  56. CSeq: 329 INVITE
  57. Server: Asterisk PBX 13.6.0
  58. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  59. Supported: replaces, timer
  60. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="229533a6"
  61. Content-Length: 0
  62.  
  63.  
  64. <------------>
  65. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  66.  
  67. <--- SIP read from UDP:192.168.1.102:5070 --->
  68. ACK sip:[email protected] SIP/2.0
  69. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1656108
  70. To: <sip:[email protected]>;tag=as253f2ec7
  71. From: "Stephen" <sip:[email protected]>;tag=8325
  72. Call-ID: [email protected]
  73. CSeq: 329 ACK
  74. Max-Forwards: 20
  75. User-Agent: NCH Software Express Talk 4.35
  76. Content-Length: 0
  77.  
  78. <------------->
  79. --- (9 headers 0 lines) ---
  80.  
  81. <--- SIP read from UDP:192.168.1.102:5070 --->
  82. INVITE sip:[email protected] SIP/2.0
  83. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1666108
  84. To: <sip:[email protected]>
  85. From: "Stephen" <sip:[email protected]>;tag=8325
  86. Call-ID: [email protected]
  87. CSeq: 330 INVITE
  88. Max-Forwards: 20
  89. User-Agent: NCH Software Express Talk 4.35
  90. Contact: <sip:[email protected]:5070>
  91. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="229533a6",uri="sip:[email protected]",response="11fa1760993dcec4f47bf1a66ac20238",opaque="",algorithm=MD5
  92. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  93. Supported: replaces
  94. Content-Type: application/sdp
  95. Content-Length: 354
  96.  
  97. v=0
  98. o=NCHSoftware-Talk 1447514436 1447514449 IN IP4 192.168.1.102
  99. s=Express Talk Call
  100. c=IN IP4 192.168.1.102
  101. t=0 0
  102. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  103. a=rtpmap:0 PCMU/8000
  104. a=rtpmap:8 PCMA/8000
  105. a=rtpmap:96 G726-32/8000
  106. a=rtpmap:3 GSM/8000
  107. a=rtpmap:13 CN/8000
  108. a=rtpmap:101 telephone-event/8000
  109. a=fmtp:101 0-16
  110. a=sendrecv
  111. a=direction:active
  112. <------------->
  113. --- (14 headers 15 lines) ---
  114. Sending to 192.168.1.102:5070 (NAT)
  115. Using INVITE request as basis request - [email protected]
  116. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  117. == Using SIP RTP CoS mark 5
  118. Found RTP audio format 0
  119. Found RTP audio format 8
  120. Found RTP audio format 96
  121. Found RTP audio format 3
  122. Found RTP audio format 13
  123. Found RTP audio format 101
  124. Found audio description format PCMU for ID 0
  125. Found audio description format PCMA for ID 8
  126. Found audio description format G726-32 for ID 96
  127. Found audio description format GSM for ID 3
  128. Found audio description format CN for ID 13
  129. Found audio description format telephone-event for ID 101
  130. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  131. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  132. Peer audio RTP is at port 192.168.1.102:8000
  133. Looking for 4183172685 in LocalSets (domain 192.168.1.207)
  134. sip_route_dump: route/path hop: <sip:[email protected]:5070>
  135.  
  136. <--- Transmitting (NAT) to 192.168.1.102:5070 --->
  137. SIP/2.0 100 Trying
  138. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1666108;received=192.168.1.102;rport=5070
  139. From: "Stephen" <sip:[email protected]>;tag=8325
  140. To: <sip:[email protected]>
  141. Call-ID: [email protected]
  142. CSeq: 330 INVITE
  143. Server: Asterisk PBX 13.6.0
  144. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  145. Supported: replaces, timer
  146. Contact: <sip:[email protected]:5060>
  147. Content-Length: 0
  148.  
  149.  
  150. <------------>
  151. -- Executing [4183172685@LocalSets:1] Log("SIP/0000FFFF0001-00000004", "NOTICE, Dialing out from "Stephen" <0000FFFF0001> to 183172685 through Foo Provider") in new stack
  152. -- Executing [4183172685@LocalSets:2] Dial("SIP/0000FFFF0001-00000004", "SIP/4183172685@callcentric") in new stack
  153. == Using SIP RTP CoS mark 5
  154. Audio is at 16966
  155. Adding codec ulaw to SDP
  156. Adding non-codec 0x1 (telephone-event) to SDP
  157. Reliably Transmitting (no NAT) to 204.11.192.171:5080:
  158. INVITE sip:[email protected] SIP/2.0
  159. Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK7f0a860b
  160. Max-Forwards: 70
  161. From: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  162. To: <sip:[email protected]>
  163. Contact: <sip:[email protected]:5060>
  164. Call-ID: [email protected]
  165. CSeq: 102 INVITE
  166. User-Agent: Asterisk PBX 13.6.0
  167. Date: Sat, 14 Nov 2015 20:23:08 GMT
  168. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  169. Supported: replaces, timer
  170. Content-Type: application/sdp
  171. Content-Length: 241
  172.  
  173. v=0
  174. o=root 1929187036 1929187036 IN IP4 192.168.1.207
  175. s=Asterisk PBX 13.6.0
  176. c=IN IP4 192.168.1.207
  177. t=0 0
  178. m=audio 16966 RTP/AVP 0 101
  179. a=rtpmap:0 PCMU/8000
  180. a=rtpmap:101 telephone-event/8000
  181. a=fmtp:101 0-16
  182. a=maxptime:150
  183. a=sendrecv
  184.  
  185. ---
  186. -- Called SIP/4183172685@callcentric
  187.  
  188. <--- SIP read from UDP:204.11.192.171:5080 --->
  189. SIP/2.0 407 Proxy Authentication Required
  190. v: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK7f0a860b;rport=58563;received=142.134.91.178
  191. f: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  192. t: <sip:[email protected]>
  193. i: [email protected]
  194. CSeq: 102 INVITE
  195. Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="c0c4886104c7d88b3f23e389761d1e4d", opaque="", stale=TRUE, algorithm=MD5
  196. l: 0
  197.  
  198. <------------->
  199. --- (8 headers 0 lines) ---
  200. Transmitting (no NAT) to 204.11.192.171:5080:
  201. ACK sip:[email protected] SIP/2.0
  202. Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK7f0a860b
  203. Max-Forwards: 70
  204. From: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  205. To: <sip:[email protected]>
  206. Contact: <sip:[email protected]:5060>
  207. Call-ID: [email protected]
  208. CSeq: 102 ACK
  209. User-Agent: Asterisk PBX 13.6.0
  210. Content-Length: 0
  211.  
  212.  
  213. ---
  214. Audio is at 16966
  215. Adding codec ulaw to SDP
  216. Adding non-codec 0x1 (telephone-event) to SDP
  217. Reliably Transmitting (no NAT) to 204.11.192.171:5080:
  218. INVITE sip:[email protected] SIP/2.0
  219. Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK6e067b8e
  220. Max-Forwards: 70
  221. From: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  222. To: <sip:[email protected]>
  223. Contact: <sip:[email protected]:5060>
  224. Call-ID: [email protected]
  225. CSeq: 103 INVITE
  226. User-Agent: Asterisk PBX 13.6.0
  227. Proxy-Authorization: Digest username="17772409788", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="c0c4886104c7d88b3f23e389761d1e4d", response="2f27d24fb40665505092a0e978f90d85"
  228. Date: Sat, 14 Nov 2015 20:23:08 GMT
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  230. Supported: replaces, timer
  231. Content-Type: application/sdp
  232. Content-Length: 241
  233.  
  234. v=0
  235. o=root 1929187036 1929187037 IN IP4 192.168.1.207
  236. s=Asterisk PBX 13.6.0
  237. c=IN IP4 192.168.1.207
  238. t=0 0
  239. m=audio 16966 RTP/AVP 0 101
  240. a=rtpmap:0 PCMU/8000
  241. a=rtpmap:101 telephone-event/8000
  242. a=fmtp:101 0-16
  243. a=maxptime:150
  244. a=sendrecv
  245.  
  246. ---
  247.  
  248. <--- SIP read from UDP:204.11.192.171:5080 --->
  249. SIP/2.0 403 Incorrect Authentication
  250. v: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK6e067b8e;rport=58563;received=142.134.91.178
  251. f: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  252. t: <sip:[email protected]>
  253. i: [email protected]
  254. CSeq: 103 INVITE
  255. l: 0
  256.  
  257. <------------->
  258. --- (7 headers 0 lines) ---
  259. Transmitting (no NAT) to 204.11.192.171:5080:
  260. ACK sip:[email protected] SIP/2.0
  261. Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK6e067b8e
  262. Max-Forwards: 70
  263. From: "Stephen" <sip:[email protected]>;tag=as5a8c7d51
  264. To: <sip:[email protected]>
  265. Contact: <sip:[email protected]:5060>
  266. Call-ID: [email protected]
  267. CSeq: 103 ACK
  268. User-Agent: Asterisk PBX 13.6.0
  269. Content-Length: 0
  270.  
  271.  
  272. ---
  273. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  274. == Everyone is busy/congested at this time (1:0/0/1)
  275. -- Executing [4183172685@LocalSets:3] PlayTones("SIP/0000FFFF0001-00000004", "congestion") in new stack
  276. -- Executing [4183172685@LocalSets:4] Hangup("SIP/0000FFFF0001-00000004", "") in new stack
  277. == Spawn extension (LocalSets, 4183172685, 4) exited non-zero on 'SIP/0000FFFF0001-00000004'
  278. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  279.  
  280. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  281. SIP/2.0 403 Forbidden
  282. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1666108;received=192.168.1.102;rport=5070
  283. From: "Stephen" <sip:[email protected]>;tag=8325
  284. To: <sip:[email protected]>;tag=as5dc2a75c
  285. Call-ID: [email protected]
  286. CSeq: 330 INVITE
  287. Server: Asterisk PBX 13.6.0
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  289. Supported: replaces, timer
  290. Content-Length: 0
  291.  
  292.  
  293. <------------>
  294.  
  295. <--- SIP read from UDP:192.168.1.102:5070 --->
  296. ACK sip:[email protected] SIP/2.0
  297. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1666108
  298. To: <sip:[email protected]>;tag=as5dc2a75c
  299. From: "Stephen" <sip:[email protected]>;tag=8325
  300. Call-ID: [email protected]
  301. CSeq: 330 ACK
  302. Max-Forwards: 20
  303. User-Agent: NCH Software Express Talk 4.35
  304. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="229533a6",uri="sip:[email protected]",response="11fa1760993dcec4f47bf1a66ac20238",opaque="",algorithm=MD5
  305. Content-Length: 0
  306.  
  307. <------------->
  308. --- (10 headers 0 lines) ---
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