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  1. *CLI> sip set debug on
  2. SIP Debugging enabled
  3. *CLI>
  4. <--- SIP read from UDP:XXXXXXXX:33895 --->
  5. INVITE sip:200@moon.light.com SIP/2.0
  6. Via: SIP/2.0/UDP XXXXXXXX:33895;rport;branch=z9hG4bKPjnm64tIzWh1vhLxCZT2eLcl3SRAucC0Xs
  7. Max-Forwards: 70
  8. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  9. To: <sip:200@moon.light.com>
  10. Contact: "Line 1" <sip:D70@XXXXXXXX:33895;ob>
  11. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  12. CSeq: 5049 INVITE
  13. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  14. Supported: replaces, 100rel, timer, norefersub
  15. Session-Expires: 1800
  16. Min-SE: 90
  17. User-Agent: Digium D70 1_1_2_0_51236
  18. Content-Type: application/sdp
  19. Content-Length: 434
  20.  
  21. v=0
  22. o=- 54187081 54187081 IN IP4 192.168.1.117
  23. s=digphn
  24. c=IN IP4 192.168.1.117
  25. t=0 0
  26. a=X-nat:0
  27. m=audio 4012 RTP/AVP 111 18 0 58 118 8 9 58 96
  28. a=rtcp:4013 IN IP4 192.168.1.117
  29. a=rtpmap:111 G726-32/8000
  30. a=rtpmap:18 G729/8000
  31. a=rtpmap:0 PCMU/8000
  32. a=rtpmap:58 L16/16000
  33. a=rtpmap:118 L16/8000
  34. a=rtpmap:8 PCMA/8000
  35. a=rtpmap:9 G722/8000
  36. a=rtpmap:58 L16-256/16000
  37. a=sendrecv
  38. a=rtpmap:96 telephone-event/8000
  39. a=fmtp:96 0-15
  40. <------------->
  41. --- (15 headers 19 lines) ---
  42. Sending to XXXXXXXX:33895 (NAT)
  43. Using INVITE request as basis request - Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  44. Found peer 'D70' for 'D70' from XXXXXXXX:33895
  45.  
  46. <--- Reliably Transmitting (NAT) to XXXXXXXX:33895 --->
  47. SIP/2.0 401 Unauthorized
  48. Via: SIP/2.0/UDP XXXXXXXX:33895;branch=z9hG4bKPjnm64tIzWh1vhLxCZT2eLcl3SRAucC0Xs;received=XXXXXXXX;rport=33895
  49. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  50. To: <sip:200@moon.light.com>;tag=as34a89324
  51. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  52. CSeq: 5049 INVITE
  53. Server: Asterisk PBX SVN-branch-1.8-r379001
  54. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  55. Supported: replaces, timer
  56. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63a8d85b"
  57. Content-Length: 0
  58.  
  59.  
  60. <------------>
  61. Scheduling destruction of SIP dialog 'Tb7TydkGjvYflJUGFp21MAshPnPsDObh' in 6400 ms (Method: INVITE)
  62.  
  63. <--- SIP read from UDP:XXXXXXXX:33895 --->
  64. ACK sip:200@moon.light.com SIP/2.0
  65. Via: SIP/2.0/UDP XXXXXXXX:33895;rport;branch=z9hG4bKPjnm64tIzWh1vhLxCZT2eLcl3SRAucC0Xs
  66. Max-Forwards: 70
  67. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  68. To: <sip:200@moon.light.com>;tag=as34a89324
  69. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  70. CSeq: 5049 ACK
  71. Content-Length: 0
  72.  
  73. <------------->
  74. --- (8 headers 0 lines) ---
  75.  
  76. <--- SIP read from UDP:XXXXXXXX:33895 --->
  77. INVITE sip:200@moon.light.com SIP/2.0
  78. Via: SIP/2.0/UDP XXXXXXXX:33895;rport;branch=z9hG4bKPjJvV65g6yauLUzHrGmkOQGiaCIkeC8HQM
  79. Max-Forwards: 70
  80. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  81. To: <sip:200@moon.light.com>
  82. Contact: "Line 1" <sip:D70@XXXXXXXX:33895;ob>
  83. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  84. CSeq: 5050 INVITE
  85. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  86. Supported: replaces, 100rel, timer, norefersub
  87. Session-Expires: 1800
  88. Min-SE: 90
  89. User-Agent: Digium D70 1_1_2_0_51236
  90. Authorization: Digest username="D70", realm="asterisk", nonce="63a8d85b", uri="sip:200@moon.light.com", response="bf8ab69a4473860a909226c6b67882a3", algorithm=MD5
  91. Content-Type: application/sdp
  92. Content-Length: 434
  93.  
  94. v=0
  95. o=- 54187081 54187081 IN IP4 192.168.1.117
  96. s=digphn
  97. c=IN IP4 192.168.1.117
  98. t=0 0
  99. a=X-nat:0
  100. m=audio 4012 RTP/AVP 111 18 0 58 118 8 9 58 96
  101. a=rtcp:4013 IN IP4 192.168.1.117
  102. a=rtpmap:111 G726-32/8000
  103. a=rtpmap:18 G729/8000
  104. a=rtpmap:0 PCMU/8000
  105. a=rtpmap:58 L16/16000
  106. a=rtpmap:118 L16/8000
  107. a=rtpmap:8 PCMA/8000
  108. a=rtpmap:9 G722/8000
  109. a=rtpmap:58 L16-256/16000
  110. a=sendrecv
  111. a=rtpmap:96 telephone-event/8000
  112. a=fmtp:96 0-15
  113. <------------->
  114. --- (16 headers 19 lines) ---
  115. Sending to XXXXXXXX:33895 (NAT)
  116. Using INVITE request as basis request - Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  117. Found peer 'D70' for 'D70' from XXXXXXXX:33895
  118. == Using SIP RTP CoS mark 5
  119. Found RTP audio format 111
  120. Found RTP audio format 18
  121. Found RTP audio format 0
  122. Found RTP audio format 58
  123. Found RTP audio format 118
  124. Found RTP audio format 8
  125. Found RTP audio format 9
  126. Found RTP audio format 58
  127. Found RTP audio format 96
  128. Found audio description format G726-32 for ID 111
  129. Found audio description format G729 for ID 18
  130. Found audio description format PCMU for ID 0
  131. Found audio description format L16 for ID 58
  132. Found audio description format L16 for ID 118
  133. Found audio description format PCMA for ID 8
  134. Found audio description format G722 for ID 9
  135. Found audio description format L16-256 for ID 58
  136. Found audio description format telephone-event for ID 96
  137. Capabilities: us - 0x4 (ulaw), peer - audio=0x994c (ulaw|alaw|g726|slin|g729|g722|slin16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  138. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  139. Peer audio RTP is at port 192.168.1.117:4012
  140. Looking for 200 in LocalSets (domain moon.light.com)
  141. list_route: hop: <sip:D70@XXXXXXXX:33895;ob>
  142.  
  143. <--- Transmitting (NAT) to XXXXXXXX:33895 --->
  144. SIP/2.0 100 Trying
  145. Via: SIP/2.0/UDP XXXXXXXX:33895;branch=z9hG4bKPjJvV65g6yauLUzHrGmkOQGiaCIkeC8HQM;received=XXXXXXXX;rport=33895
  146. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  147. To: <sip:200@moon.light.com>
  148. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  149. CSeq: 5050 INVITE
  150. Server: Asterisk PBX SVN-branch-1.8-r379001
  151. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  152. Supported: replaces, timer
  153. Session-Expires: 1800;refresher=uas
  154. Contact: <sip:200@YYYYYYYYYY:5060>
  155. Content-Length: 0
  156.  
  157.  
  158. Audio is at 28820
  159. Adding codec 0x4 (ulaw) to SDP
  160. Adding non-codec 0x1 (telephone-event) to SDP
  161.  
  162. <--- Reliably Transmitting (NAT) to XXXXXXXX:33895 --->
  163. SIP/2.0 200 OK
  164. Via: SIP/2.0/UDP XXXXXXXX:33895;branch=z9hG4bKPjJvV65g6yauLUzHrGmkOQGiaCIkeC8HQM;received=XXXXXXXX;rport=33895
  165. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  166. To: <sip:200@moon.light.com>;tag=as0854619c
  167. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  168. CSeq: 5050 INVITE
  169. Server: Asterisk PBX SVN-branch-1.8-r379001
  170. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  171. Supported: replaces, timer
  172. Session-Expires: 1800;refresher=uas
  173. Contact: <sip:200@YYYYYYYYYY:5060>
  174. Content-Type: application/sdp
  175. Require: timer
  176. Content-Length: 279
  177.  
  178. v=0
  179. o=root 1792503678 1792503678 IN IP4 YYYYYYYYYY
  180. s=Asterisk PBX SVN-branch-1.8-r379001
  181. c=IN IP4 YYYYYYYYYY
  182. t=0 0
  183. m=audio 28820 RTP/AVP 0 96
  184. a=rtpmap:0 PCMU/8000
  185. a=rtpmap:96 telephone-event/8000
  186. a=fmtp:96 0-16
  187. a=silenceSupp:off - - - -
  188. a=ptime:20
  189. a=sendrecv
  190.  
  191. <------------>
  192. Retransmitting #1 (NAT) to XXXXXXXX:33895:
  193. SIP/2.0 200 OK
  194. Via: SIP/2.0/UDP XXXXXXXX:33895;branch=z9hG4bKPjJvV65g6yauLUzHrGmkOQGiaCIkeC8HQM;received=XXXXXXXX;rport=33895
  195. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  196. To: <sip:200@moon.light.com>;tag=as0854619c
  197. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  198. CSeq: 5050 INVITE
  199. Server: Asterisk PBX SVN-branch-1.8-r379001
  200. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  201. Supported: replaces, timer
  202. Session-Expires: 1800;refresher=uas
  203. Contact: <sip:200@YYYYYYYYYY:5060>
  204. Content-Type: application/sdp
  205. Require: timer
  206. Content-Length: 279
  207.  
  208. v=0
  209. o=root 1792503678 1792503678 IN IP4 YYYYYYYYYY
  210. s=Asterisk PBX SVN-branch-1.8-r379001
  211. c=IN IP4 YYYYYYYYYY
  212. t=0 0
  213. m=audio 28820 RTP/AVP 0 96
  214. a=rtpmap:0 PCMU/8000
  215. a=rtpmap:96 telephone-event/8000
  216. a=fmtp:96 0-16
  217. a=silenceSupp:off - - - -
  218. a=ptime:20
  219. a=sendrecv
  220.  
  221. ---
  222.  
  223. <--- SIP read from UDP:XXXXXXXX:33895 --->
  224. ACK sip:200@YYYYYYYYYY:5060 SIP/2.0
  225. Via: SIP/2.0/UDP XXXXXXXX:33895;rport;branch=z9hG4bKPj-7TqmGmdW7D6j708dNohTIfYBE7yaV4t
  226. Max-Forwards: 70
  227. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  228. To: <sip:200@moon.light.com>;tag=as0854619c
  229. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  230. CSeq: 5050 ACK
  231. Content-Length: 0
  232.  
  233. <------------->
  234. --- (8 headers 0 lines) ---
  235.  
  236. <--- SIP read from UDP:XXXXXXXX:33895 --->
  237. ACK sip:200@YYYYYYYYYY:5060 SIP/2.0
  238. Via: SIP/2.0/UDP XXXXXXXX:33895;rport;branch=z9hG4bKPj-7TqmGmdW7D6j708dNohTIfYBE7yaV4t
  239. Max-Forwards: 70
  240. From: "Line 1" <sip:D70@moon.light.com>;tag=K4NWoy4W6HrQMPKIi-wRiEIak.cksB38
  241. To: <sip:200@moon.light.com>;tag=as0854619c
  242. Call-ID: Tb7TydkGjvYflJUGFp21MAshPnPsDObh
  243. CSeq: 5050 ACK
  244. Content-Length: 0
  245.  
  246. <------------->
  247. --- (8 headers 0 lines) ---
  248. -- Executing [s@CallerOK:7] Wait("SIP/D70-00000000", "1") in new stack
  249. -- Executing [s@CallerOK:8] Read("SIP/D70-00000000", "ETA,access-granted,1,,,5") in new stack
  250. -- Accepting a maximum of 1 digits.
  251. -- <SIP/D70-00000000> Playing 'access-granted.ulaw' (language 'en')
  252. sip set debug off
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