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Asterisk 2011-11-01

Nov 1st, 2011
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  1.  
  2. athomehost*CLI> sip set debug on
  3. SIP Debugging enabled
  4. Reliably Transmitting (NAT) to 192.168.8.110:5060:
  5. OPTIONS sip:101@192.168.8.110:5060 SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5bb9c4f0;rport
  7. Max-Forwards: 70
  8. From: "asterisk" <sip:asterisk@192.168.8.1>;tag=as3e586d83
  9. To: <sip:101@192.168.8.110:5060>
  10. Contact: <sip:asterisk@192.168.8.1:5060>
  11. Call-ID: 339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060
  12. CSeq: 102 OPTIONS
  13. User-Agent: Asterisk PBX 1.8.4
  14. Date: Tue, 01 Nov 2011 15:48:30 GMT
  15. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  16. Supported: replaces
  17. Content-Length: 0
  18.  
  19.  
  20. ---
  21.  
  22. <--- SIP read from UDP:192.168.8.110:5060 --->
  23. SIP/2.0 486 Busy Here
  24. To: <sip:101@192.168.8.110:5060>;tag=a46b2f48cb3e8cb9i1
  25. From: "asterisk" <sip:asterisk@192.168.8.1>;tag=as3e586d83
  26. Call-ID: 339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060
  27. CSeq: 102 OPTIONS
  28. Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5bb9c4f0
  29. Server: Linksys/SPA2102-5.1.5(a)
  30. Content-Length: 0
  31. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  32. Supported: x-sipura, replaces
  33.  
  34. <------------->
  35. --- (10 headers 0 lines) ---
  36. Really destroying SIP dialog '339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060' Method: OPTIONS
  37.  
  38. <--- SIP read from UDP:192.168.8.110:5060 --->
  39. INVITE sip:17772022880@192.168.8.1 SIP/2.0
  40. Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a
  41. From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
  42. To: <sip:17772022880@192.168.8.1>
  43. Remote-Party-ID: L2SPA2102 <sip:101@192.168.8.1>;screen=yes;party=calling
  44. Call-ID: d65495ce-e517150a@192.168.8.110
  45. CSeq: 101 INVITE
  46. Max-Forwards: 70
  47. Contact: L2SPA2102 <sip:101@192.168.8.110:5060>
  48. Expires: 240
  49. User-Agent: Linksys/SPA2102-5.1.5(a)
  50. Content-Length: 446
  51. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  52. Supported: x-sipura, replaces
  53. Content-Type: application/sdp
  54.  
  55. v=0
  56. o=- 5320193 5320193 IN IP4 192.168.8.110
  57. s=-
  58. c=IN IP4 192.168.8.110
  59. t=0 0
  60. m=audio 16430 RTP/AVP 18 0 2 4 8 96 97 98 100 101
  61. a=rtpmap:18 G729a/8000
  62. a=rtpmap:0 PCMU/8000
  63. a=rtpmap:2 G726-32/8000
  64. a=rtpmap:4 G723/8000
  65. a=rtpmap:8 PCMA/8000
  66. a=rtpmap:96 G726-40/8000
  67. a=rtpmap:97 G726-24/8000
  68. a=rtpmap:98 G726-16/8000
  69. a=rtpmap:100 NSE/8000
  70. a=fmtp:100 192-193
  71. a=rtpmap:101 telephone-event/8000
  72. a=fmtp:101 0-15
  73. a=ptime:30
  74. a=sendrecv
  75. <------------->
  76. --- (15 headers 20 lines) ---
  77. Sending to 192.168.8.110:5060 (no NAT)
  78. Using INVITE request as basis request - d65495ce-e517150a@192.168.8.110
  79. Found peer '101' for '101' from 192.168.8.110:5060
  80. Found RTP audio format 18
  81. Found RTP audio format 0
  82. Found RTP audio format 2
  83. Found RTP audio format 4
  84. Found RTP audio format 8
  85. Found RTP audio format 96
  86. Found RTP audio format 97
  87. Found RTP audio format 98
  88. Found RTP audio format 100
  89. Found RTP audio format 101
  90. Found audio description format G729a for ID 18
  91. Found audio description format PCMU for ID 0
  92. Found audio description format G726-32 for ID 2
  93. Found audio description format G723 for ID 4
  94. Found audio description format PCMA for ID 8
  95. Found audio description format G726-40 for ID 96
  96. Found audio description format G726-24 for ID 97
  97. Found audio description format G726-16 for ID 98
  98. Found audio description format NSE for ID 100
  99. Found audio description format telephone-event for ID 101
  100. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  101. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  102. Peer audio RTP is at port 192.168.8.110:16430
  103. Looking for 17772022880 in mario-default (domain 192.168.8.1)
  104. list_route: hop: <sip:101@192.168.8.110:5060>
  105.  
  106. <--- Transmitting (NAT) to 192.168.8.110:5060 --->
  107. SIP/2.0 100 Trying
  108. Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
  109. From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
  110. To: <sip:17772022880@192.168.8.1>
  111. Call-ID: d65495ce-e517150a@192.168.8.110
  112. CSeq: 101 INVITE
  113. Server: Asterisk PBX 1.8.4
  114. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  115. Supported: replaces
  116. Contact: <sip:17772022880@192.168.8.1:5060>
  117. Content-Length: 0
  118.  
  119.  
  120. <------------>
  121.  
  122. <--- Transmitting (NAT) to 192.168.8.110:5060 --->
  123. SIP/2.0 180 Ringing
  124. Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
  125. From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
  126. To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
  127. Call-ID: d65495ce-e517150a@192.168.8.110
  128. CSeq: 101 INVITE
  129. Server: Asterisk PBX 1.8.4
  130. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  131. Supported: replaces
  132. Contact: <sip:17772022880@192.168.8.1:5060>
  133. Content-Length: 0
  134.  
  135.  
  136. <------------>
  137.  
  138. <--- Reliably Transmitting (NAT) to 192.168.8.110:5060 --->
  139. SIP/2.0 503 Service Unavailable
  140. Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
  141. From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
  142. To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
  143. Call-ID: d65495ce-e517150a@192.168.8.110
  144. CSeq: 101 INVITE
  145. Server: Asterisk PBX 1.8.4
  146. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  147. Supported: replaces
  148. Content-Length: 0
  149.  
  150.  
  151. <------------>
  152.  
  153. <--- SIP read from UDP:192.168.8.110:5060 --->
  154. ACK sip:17772022880@192.168.8.1 SIP/2.0
  155. Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a
  156. From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
  157. To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
  158. Call-ID: d65495ce-e517150a@192.168.8.110
  159. CSeq: 101 ACK
  160. Max-Forwards: 70
  161. Contact: L2SPA2102 <sip:101@192.168.8.110:5060>
  162. User-Agent: Linksys/SPA2102-5.1.5(a)
  163. Content-Length: 0
  164.  
  165. <------------->
  166. --- (10 headers 0 lines) ---
  167. Really destroying SIP dialog 'd65495ce-e517150a@192.168.8.110' Method: ACK
  168. Really destroying SIP dialog '28a33b8611bed2be40140e0b4858ae2b@[c0a8:801:e82a:c97f::]' Method: REGISTER
  169. athomehost*CLI>
  170.  
  171.  
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