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Jan 10th, 2017
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  1. ➜ ~ sudo asterisk -rvvvvvv
  2. Asterisk 11.13.1~dfsg-2+deb8u1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 11.13.1~dfsg-2+deb8u1 currently running on david (pid = 29264)
  10.  
  11. <--- SIP read from UDP:192.168.0.6:5062 --->
  12. INVITE sip:101@192.168.0.2:5060 SIP/2.0
  13. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK519449704;rport
  14. Route: <sip:192.168.0.2:5060;lr>
  15. From: <sip:FXO@192.168.0.2>;tag=1669346963
  16. To: <sip:101@192.168.0.2:5060>
  17. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  18. CSeq: 10 INVITE
  19. Contact: <sip:FXO@192.168.0.6:5062>
  20. Max-Forwards: 70
  21. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  22. Supported: replaces, path, timer, eventlist
  23. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  24. Content-Type: application/sdp
  25. Accept: application/sdp, application/dtmf-relay
  26. Content-Length: 384
  27.  
  28. v=0
  29. o=FXO 8002 8000 IN IP4 192.168.0.6
  30. s=SIP Call
  31. c=IN IP4 192.168.0.6
  32. t=0 0
  33. m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100
  34. a=sendrecv
  35. a=rtpmap:0 PCMU/8000
  36. a=ptime:20
  37. a=rtpmap:8 PCMA/8000
  38. a=rtpmap:4 G723/8000
  39. a=rtpmap:18 G729/8000
  40. a=fmtp:18 annexb=no
  41. a=rtpmap:2 G726-32/8000
  42. a=rtpmap:97 iLBC/8000
  43. a=fmtp:97 mode=20
  44. a=rtpmap:102 G729E/8000
  45. a=rtpmap:100 AAL2-G726-16/8000
  46. <------------->
  47. --- (15 headers 18 lines) ---
  48. Sending to 192.168.0.6:5062 (no NAT)
  49. Sending to 192.168.0.6:5062 (no NAT)
  50. Using INVITE request as basis request - 1430638416-5062-2@BJC.BGI.A.G
  51. Found peer 'FXO' for 'FXO' from 192.168.0.6:5062
  52.  
  53. <--- Reliably Transmitting (no NAT) to 192.168.0.6:5062 --->
  54. SIP/2.0 401 Unauthorized
  55. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK519449704;received=192.168.0.6;rport=5062
  56. From: <sip:FXO@192.168.0.2>;tag=1669346963
  57. To: <sip:101@192.168.0.2:5060>;tag=as1f672c5a
  58. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  59. CSeq: 10 INVITE
  60. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
  61. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  62. Supported: replaces, timer
  63. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4eaef046"
  64. Content-Length: 0
  65.  
  66.  
  67. <------------>
  68. Scheduling destruction of SIP dialog '1430638416-5062-2@BJC.BGI.A.G' in 6400 ms (Method: INVITE)
  69.  
  70. <--- SIP read from UDP:192.168.0.6:5062 --->
  71. ACK sip:101@192.168.0.2:5060 SIP/2.0
  72. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK519449704;rport
  73. Route: <sip:192.168.0.2:5060;lr>
  74. From: <sip:FXO@192.168.0.2>;tag=1669346963
  75. To: <sip:101@192.168.0.2:5060>;tag=as1f672c5a
  76. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  77. CSeq: 10 ACK
  78. Content-Length: 0
  79.  
  80. <------------->
  81. --- (8 headers 0 lines) ---
  82.  
  83. <--- SIP read from UDP:192.168.0.6:5062 --->
  84. INVITE sip:101@192.168.0.2:5060 SIP/2.0
  85. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1463688355;rport
  86. Route: <sip:192.168.0.2:5060;lr>
  87. From: <sip:FXO@192.168.0.2>;tag=1669346963
  88. To: <sip:101@192.168.0.2:5060>
  89. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  90. CSeq: 11 INVITE
  91. Contact: <sip:FXO@192.168.0.6:5062>
  92. Authorization: Digest username="FXO", realm="asterisk", nonce="4eaef046", uri="sip:101@192.168.0.2:5060", response="1c510e536a04113210af178f9f2fde5d", algorithm=MD5
  93. Max-Forwards: 70
  94. User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
  95. Supported: replaces, path, timer, eventlist
  96. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  97. Content-Type: application/sdp
  98. Accept: application/sdp, application/dtmf-relay
  99. Content-Length: 384
  100.  
  101. v=0
  102. o=FXO 8002 8000 IN IP4 192.168.0.6
  103. s=SIP Call
  104. c=IN IP4 192.168.0.6
  105. t=0 0
  106. m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100
  107. a=sendrecv
  108. a=rtpmap:0 PCMU/8000
  109. a=ptime:20
  110. a=rtpmap:8 PCMA/8000
  111. a=rtpmap:4 G723/8000
  112. a=rtpmap:18 G729/8000
  113. a=fmtp:18 annexb=no
  114. a=rtpmap:2 G726-32/8000
  115. a=rtpmap:97 iLBC/8000
  116. a=fmtp:97 mode=20
  117. a=rtpmap:102 G729E/8000
  118. a=rtpmap:100 AAL2-G726-16/8000
  119. <------------->
  120. --- (16 headers 18 lines) ---
  121. Sending to 192.168.0.6:5062 (no NAT)
  122. Using INVITE request as basis request - 1430638416-5062-2@BJC.BGI.A.G
  123. Found peer 'FXO' for 'FXO' from 192.168.0.6:5062
  124. == Using SIP RTP CoS mark 5
  125. Found RTP audio format 0
  126. Found RTP audio format 8
  127. Found RTP audio format 4
  128. Found RTP audio format 18
  129. Found RTP audio format 2
  130. Found RTP audio format 97
  131. Found RTP audio format 102
  132. Found RTP audio format 100
  133. Found audio description format PCMU for ID 0
  134. Found audio description format PCMA for ID 8
  135. Found audio description format G723 for ID 4
  136. Found audio description format G729 for ID 18
  137. Found audio description format G726-32 for ID 2
  138. Found audio description format iLBC for ID 97
  139. Found unknown media description format G729E for ID 102
  140. Found unknown media description format AAL2-G726-16 for ID 100
  141. Capabilities: us - (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
  142. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  143. Peer audio RTP is at port 192.168.0.6:5013
  144. Looking for 101 in phones (domain 192.168.0.2)
  145. list_route: hop: <sip:FXO@192.168.0.6:5062>
  146.  
  147. <--- Transmitting (no NAT) to 192.168.0.6:5062 --->
  148. SIP/2.0 100 Trying
  149. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1463688355;received=192.168.0.6;rport=5062
  150. From: <sip:FXO@192.168.0.2>;tag=1669346963
  151. To: <sip:101@192.168.0.2:5060>
  152. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  153. CSeq: 11 INVITE
  154. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
  155. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  156. Supported: replaces, timer
  157. Session-Expires: 1800;refresher=uas
  158. Contact: <sip:101@192.168.0.2:5060>
  159. Content-Length: 0
  160.  
  161.  
  162. <------------>
  163. -- Executing [101@phones:1] Dial("SIP/FXO-0000004c", "SIP/101") in new stack
  164. == Using SIP RTP CoS mark 5
  165. Audio is at 15406
  166. Adding codec 100003 (ulaw) to SDP
  167. Adding codec 100004 (alaw) to SDP
  168. Adding codec 100002 (gsm) to SDP
  169. Adding codec 100017 (testlaw) to SDP
  170. Adding non-codec 0x1 (telephone-event) to SDP
  171. Reliably Transmitting (no NAT) to 192.168.0.101:5060:
  172. INVITE sip:101@192.168.0.101:5060 SIP/2.0
  173. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK37ea183d
  174. Max-Forwards: 70
  175. From: <sip:FXO@192.168.0.2>;tag=as4bb3a04b
  176. To: <sip:101@192.168.0.101:5060>
  177. Contact: <sip:FXO@192.168.0.2:5060>
  178. Call-ID: 35f456243ed30b9978cf257b6a60d02b@192.168.0.2:5060
  179. CSeq: 102 INVITE
  180. User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
  181. Date: Tue, 10 Jan 2017 19:06:43 GMT
  182. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  183. Supported: replaces, timer
  184. Content-Type: application/sdp
  185. Content-Length: 295
  186.  
  187. v=0
  188. o=root 1412127001 1412127001 IN IP4 192.168.0.2
  189. s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
  190. c=IN IP4 192.168.0.2
  191. t=0 0
  192. m=audio 15406 RTP/AVP 0 8 3 101
  193. a=rtpmap:0 PCMU/8000
  194. a=rtpmap:8 PCMA/8000
  195. a=rtpmap:3 GSM/8000
  196. a=rtpmap:101 telephone-event/8000
  197. a=fmtp:101 0-16
  198. a=ptime:20
  199. a=sendrecv
  200.  
  201. ---
  202. -- Called SIP/101
  203.  
  204. <--- SIP read from UDP:192.168.0.101:5060 --->
  205. SIP/2.0 100 Trying
  206. To: <sip:101@192.168.0.101:5060>
  207. From: <sip:FXO@192.168.0.2>;tag=as4bb3a04b
  208. Call-ID: 35f456243ed30b9978cf257b6a60d02b@192.168.0.2:5060
  209. CSeq: 102 INVITE
  210. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK37ea183d
  211. Server: Linksys/SPA942-6.1.5(a)
  212. Content-Length: 0
  213.  
  214. <------------->
  215. --- (8 headers 0 lines) ---
  216.  
  217. <--- SIP read from UDP:192.168.0.101:5060 --->
  218. SIP/2.0 180 Ringing
  219. To: <sip:101@192.168.0.101:5060>;tag=7c11d4cd6cb5e63ei0
  220. From: <sip:FXO@192.168.0.2>;tag=as4bb3a04b
  221. Call-ID: 35f456243ed30b9978cf257b6a60d02b@192.168.0.2:5060
  222. CSeq: 102 INVITE
  223. Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK37ea183d
  224. Contact: "Vlatko" <sip:101@192.168.0.101:5060>
  225. Server: Linksys/SPA942-6.1.5(a)
  226. Content-Length: 0
  227.  
  228. <------------->
  229. --- (9 headers 0 lines) ---
  230. list_route: hop: <sip:101@192.168.0.101:5060>
  231. -- SIP/101-0000004d is ringing
  232.  
  233. <--- Transmitting (no NAT) to 192.168.0.6:5062 --->
  234. SIP/2.0 180 Ringing
  235. Via: SIP/2.0/UDP 192.168.0.6:5062;branch=z9hG4bK1463688355;received=192.168.0.6;rport=5062
  236. From: <sip:FXO@192.168.0.2>;tag=1669346963
  237. To: <sip:101@192.168.0.2:5060>;tag=as76de12a0
  238. Call-ID: 1430638416-5062-2@BJC.BGI.A.G
  239. CSeq: 11 INVITE
  240. Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  242. Supported: replaces, timer
  243. Session-Expires: 1800;refresher=uas
  244. Contact: <sip:101@192.168.0.2:5060>
  245. Content-Length: 0
  246.  
  247.  
  248. <------------>
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