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  1. *CLI> sip set debug on
  2. SIP Debugging enabled
  3. *CLI>
  4. <--- SIP read from UDP:127.0.0.1:60384 --->
  5. INVITE sip:100@127.0.0.1 SIP/2.0
  6. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-8f16615c64166430-1---d8754z-;rport
  7. Max-Forwards: 70
  8. Contact: <sip:6001@127.0.0.1:60384>
  9. To: <sip:100@127.0.0.1>
  10. From: <sip:6001@127.0.0.1>;tag=743f210b
  11. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  12. CSeq: 1 INVITE
  13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  14. Content-Type: application/sdp
  15. Supported: replaces
  16. User-Agent: Bria 3 release 3.5.5 stamp 71243
  17. Content-Length: 254
  18.  
  19. v=0
  20. o=- 1395337668422076 1 IN IP4 127.0.0.1
  21. s=Bria 3 release 3.5.5 stamp 71243
  22. c=IN IP4 127.0.0.1
  23. t=0 0
  24. m=audio 61608 RTP/AVP 9 8 0 18 101
  25. a=rtpmap:18 G729/8000
  26. a=fmtp:18 annexb=yes
  27. a=rtpmap:101 telephone-event/8000
  28. a=fmtp:101 0-15
  29. a=sendrecv
  30. <------------->
  31. --- (13 headers 11 lines) ---
  32. Sending to 127.0.0.1:60384 (no NAT)
  33. Sending to 127.0.0.1:60384 (no NAT)
  34. Using INVITE request as basis request - ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  35. Found peer '6001' for '6001' from 127.0.0.1:60384
  36.  
  37. <--- Reliably Transmitting (no NAT) to 127.0.0.1:60384 --->
  38. SIP/2.0 401 Unauthorized
  39. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-8f16615c64166430-1---d8754z-;received=127.0.0.1;rport=60384
  40. From: <sip:6001@127.0.0.1>;tag=743f210b
  41. To: <sip:100@127.0.0.1>;tag=as22a94546
  42. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  43. CSeq: 1 INVITE
  44. Server: Asterisk PBX 12.1.1
  45. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  46. Supported: replaces, timer
  47. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="117a8949"
  48. Content-Length: 0
  49.  
  50.  
  51. <------------>
  52. Scheduling destruction of SIP dialog 'ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg' in 32000 ms (Method: INVITE)
  53.  
  54. <--- SIP read from UDP:127.0.0.1:60384 --->
  55. ACK sip:100@127.0.0.1 SIP/2.0
  56. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-8f16615c64166430-1---d8754z-;rport
  57. Max-Forwards: 70
  58. To: <sip:100@127.0.0.1>;tag=as22a94546
  59. From: <sip:6001@127.0.0.1>;tag=743f210b
  60. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  61. CSeq: 1 ACK
  62. Content-Length: 0
  63.  
  64. <------------->
  65. --- (8 headers 0 lines) ---
  66.  
  67. <--- SIP read from UDP:127.0.0.1:60384 --->
  68. INVITE sip:100@127.0.0.1 SIP/2.0
  69. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-2607243247531f62-1---d8754z-;rport
  70. Max-Forwards: 70
  71. Contact: <sip:6001@127.0.0.1:60384>
  72. To: <sip:100@127.0.0.1>
  73. From: <sip:6001@127.0.0.1>;tag=743f210b
  74. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  75. CSeq: 2 INVITE
  76. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  77. Content-Type: application/sdp
  78. Supported: replaces
  79. User-Agent: Bria 3 release 3.5.5 stamp 71243
  80. Authorization: Digest username="6001",realm="asterisk",nonce="117a8949",uri="sip:100@127.0.0.1",response="274f52d646fb0e3f713c7b4ff064c4ea",algorithm=MD5
  81. Content-Length: 254
  82.  
  83. v=0
  84. o=- 1395337668422076 1 IN IP4 127.0.0.1
  85. s=Bria 3 release 3.5.5 stamp 71243
  86. c=IN IP4 127.0.0.1
  87. t=0 0
  88. m=audio 61608 RTP/AVP 9 8 0 18 101
  89. a=rtpmap:18 G729/8000
  90. a=fmtp:18 annexb=yes
  91. a=rtpmap:101 telephone-event/8000
  92. a=fmtp:101 0-15
  93. a=sendrecv
  94. <------------->
  95. --- (14 headers 11 lines) ---
  96. Sending to 127.0.0.1:60384 (no NAT)
  97. Using INVITE request as basis request - ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  98. Found peer '6001' for '6001' from 127.0.0.1:60384
  99. Found RTP audio format 9
  100. Found RTP audio format 8
  101. Found RTP audio format 0
  102. Found RTP audio format 18
  103. Found RTP audio format 101
  104. Found audio description format G729 for ID 18
  105. Found audio description format telephone-event for ID 101
  106. Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw)
  107. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  108. Peer audio RTP is at port 127.0.0.1:61608
  109. Looking for 100 in from-internal (domain 127.0.0.1)
  110. list_route: route/path hop: <sip:6001@127.0.0.1:60384>
  111.  
  112. <--- Transmitting (no NAT) to 127.0.0.1:60384 --->
  113. SIP/2.0 100 Trying
  114. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-2607243247531f62-1---d8754z-;received=127.0.0.1;rport=60384
  115. From: <sip:6001@127.0.0.1>;tag=743f210b
  116. To: <sip:100@127.0.0.1>
  117. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  118. CSeq: 2 INVITE
  119. Server: Asterisk PBX 12.1.1
  120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  121. Supported: replaces, timer
  122. Contact: <sip:100@127.0.0.1:5060>
  123. Content-Length: 0
  124.  
  125.  
  126. <------------>
  127. -- Executing [100@from-internal:1] Answer("SIP/6001-00000002", "") in new stack
  128. Audio is at 12846
  129. Adding codec 100003 (ulaw) to SDP
  130. Adding non-codec 0x1 (telephone-event) to SDP
  131.  
  132. <--- Reliably Transmitting (no NAT) to 127.0.0.1:60384 --->
  133. SIP/2.0 200 OK
  134. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-2607243247531f62-1---d8754z-;received=127.0.0.1;rport=60384
  135. From: <sip:6001@127.0.0.1>;tag=743f210b
  136. To: <sip:100@127.0.0.1>;tag=as364ced19
  137. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  138. CSeq: 2 INVITE
  139. Server: Asterisk PBX 12.1.1
  140. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  141. Supported: replaces, timer
  142. Contact: <sip:100@127.0.0.1:5060>
  143. Content-Type: application/sdp
  144. Content-Length: 272
  145.  
  146. v=0
  147. o=root 1811711720 1811711720 IN IP4 127.0.0.1
  148. s=Asterisk PBX 12.1.1
  149. c=IN IP4 127.0.0.1
  150. t=0 0
  151. m=audio 12846 RTP/AVP 0 101
  152. a=rtpmap:0 PCMU/8000
  153. a=rtpmap:101 telephone-event/8000
  154. a=fmtp:101 0-16
  155. a=silenceSupp:off - - - -
  156. a=ptime:20
  157. a=maxptime:150
  158. a=sendrecv
  159.  
  160. <------------>
  161.  
  162. <--- SIP read from UDP:127.0.0.1:60384 --->
  163. ACK sip:100@127.0.0.1:5060 SIP/2.0
  164. Via: SIP/2.0/UDP 127.0.0.1:60384;branch=z9hG4bK-d8754z-4f8b4277e4fd0354-1---d8754z-;rport
  165. Max-Forwards: 70
  166. Contact: <sip:6001@127.0.0.1:60384>
  167. To: <sip:100@127.0.0.1>;tag=as364ced19
  168. From: <sip:6001@127.0.0.1>;tag=743f210b
  169. Call-ID: ZGU3MDlhNWFiZGM2YmZmMTQwMTU0YWYyMWI0MWI5MTg
  170. CSeq: 2 ACK
  171. User-Agent: Bria 3 release 3.5.5 stamp 71243
  172. Content-Length: 0
  173.  
  174. <------------->
  175. --- (10 headers 0 lines) ---
  176. > 0x10204c800 -- Probation passed - setting RTP source address to 127.0.0.1:61608
  177. -- Executing [100@from-internal:2] Wait("SIP/6001-00000002", "1") in new stack
  178. -- Executing [100@from-internal:3] Playback("SIP/6001-00000002", "hello-world") in new stack
  179. -- <SIP/6001-00000002> Playing 'hello-world.ulaw' (language 'en')
  180.  
  181. <--- SIP read from UDP:127.0.0.1:60384 --->
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