SIP Dump

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  1. <--- SIP read from UDP:10.226.31.1:5060 --->
  2. INVITE sip:4401@10.225.172.241:5060;user=phone SIP/2.0
  3. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c
  4. To: <sip:4401@10.225.172.241>
  5. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  6. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  7. CSeq: 1760593249 INVITE
  8. Max-Forwards: 8
  9. Contact: <sip:98475330@10.226.31.1:5060>
  10. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  11. Supported: 100rel,timer
  12. P-Charging-Vector: icid-value=00710-2016121510413
  13. User-Agent: ZTE Softswitch/1.0.0
  14. Session-Expires: 1800
  15. Min-SE: 1800
  16. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
  17. Content-Type: application/sdp
  18. Content-Length: 182
  19.  
  20. v=0
  21. o=ZTE 10 909382083 IN IP4 10.226.31.1
  22. s=phone-call
  23. c=IN IP4 10.226.31.1
  24. t=0 0
  25. m=audio 38224 RTP/AVP 0 8 101
  26. a=rtpmap:101 telephone-event/8000
  27. a=fmtp:101 0-15
  28. a=ptime:20
  29. <------------->
  30. --- (17 headers 9 lines) ---
  31. Sending to 10.226.31.1:5060 (NAT)
  32. Sending to 10.226.31.1:5060 (NAT)
  33. Using INVITE request as basis request - 587f800252b12a39-0292-2076@172.19.30.1
  34. Found peer 'SIP_Incoming' for '98475330' from 10.226.31.1:5060
  35.   == Using SIP RTP TOS bits 184
  36.   == Using SIP RTP CoS mark 5
  37. Found RTP audio format 0
  38. Found RTP audio format 8
  39. Found RTP audio format 101
  40. Found audio description format telephone-event for ID 101
  41. Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
  42. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  43. Peer audio RTP is at port 10.226.31.1:38224
  44. Looking for 4401 in from-trunk (domain 10.225.172.241)
  45. sip_route_dump: route/path hop: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  46.  
  47. <--- Transmitting (NAT) to 10.226.31.1:5060 --->
  48. SIP/2.0 100 Trying
  49. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c;received=10.226.31.1;rport=5060
  50. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  51. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  52. To: <sip:4401@10.225.172.241>
  53. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  54. CSeq: 1760593249 INVITE
  55. Server: FPBX-13.0.190.7(13.12.2)
  56. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  57. Supported: replaces, timer
  58. Session-Expires: 1800;refresher=uas
  59. Contact: <sip:4401@10.226.31.1:5060>
  60. Content-Length: 0
  61.  
  62.  
  63. <------------>
  64.     -- Executing [4401@from-trunk:1] Set("SIP/SIP_Incoming-0000001b", "__FROM_DID=4401") in new stack
  65.     -- Executing [4401@from-trunk:2] NoOp("SIP/SIP_Incoming-0000001b", "Received an unknown call with DID set to 4401") in new stack
  66.     -- Executing [4401@from-trunk:3] Goto("SIP/SIP_Incoming-0000001b", "s,a2") in new stack
  67.     -- Goto (from-trunk,s,2)
  68.     -- Executing [s@from-trunk:2] Answer("SIP/SIP_Incoming-0000001b", "") in new stack
  69. Audio is at 15036
  70. Adding codec alaw to SDP
  71. Adding codec ulaw to SDP
  72. Adding non-codec 0x1 (telephone-event) to SDP
  73.  
  74. <--- Reliably Transmitting (NAT) to 10.226.31.1:5060 --->
  75. SIP/2.0 200 OK
  76. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c;received=10.226.31.1;rport=5060
  77. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  78. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  79. To: <sip:4401@10.225.172.241>;tag=as6abe70aa
  80. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  81. CSeq: 1760593249 INVITE
  82. Server: FPBX-13.0.190.7(13.12.2)
  83. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  84. Supported: replaces, timer
  85. Session-Expires: 1800;refresher=uas
  86. Contact: <sip:4401@10.226.31.1:5060>
  87. Content-Type: application/sdp
  88. Require: timer
  89. Content-Length: 272
  90.  
  91. v=0
  92. o=root 720334440 720334440 IN IP4 10.226.31.1
  93. s=Asterisk PBX 13.12.2
  94. c=IN IP4 10.226.31.1
  95. t=0 0
  96. m=audio 15036 RTP/AVP 8 0 101
  97. a=rtpmap:8 PCMA/8000
  98. a=rtpmap:0 PCMU/8000
  99. a=rtpmap:101 telephone-event/8000
  100. a=fmtp:101 0-16
  101. a=ptime:20
  102. a=maxptime:150
  103. a=sendrecv
  104.  
  105. <------------>
  106. Retransmitting #1 (NAT) to 10.226.31.1:5060:
  107. SIP/2.0 200 OK
  108. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c;received=10.226.31.1;rport=5060
  109. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  110. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  111. To: <sip:4401@10.225.172.241>;tag=as6abe70aa
  112. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  113. CSeq: 1760593249 INVITE
  114. Server: FPBX-13.0.190.7(13.12.2)
  115. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  116. Supported: replaces, timer
  117. Session-Expires: 1800;refresher=uas
  118. Contact: <sip:4401@10.226.31.1:5060>
  119. Content-Type: application/sdp
  120. Require: timer
  121. Content-Length: 272
  122.  
  123. v=0
  124. o=root 720334440 720334440 IN IP4 10.226.31.1
  125. s=Asterisk PBX 13.12.2
  126. c=IN IP4 10.226.31.1
  127. t=0 0
  128. m=audio 15036 RTP/AVP 8 0 101
  129. a=rtpmap:8 PCMA/8000
  130. a=rtpmap:0 PCMU/8000
  131. a=rtpmap:101 telephone-event/8000
  132. a=fmtp:101 0-16
  133. a=ptime:20
  134. a=maxptime:150
  135. a=sendrecv
  136.  
  137. ---
  138.     -- Executing [s@from-trunk:3] Log("SIP/SIP_Incoming-0000001b", "WARNING,Friendly Scanner from 10.226.31.1") in new stack
  139. [2016-12-15 10:53:55] WARNING[5321][C-00000011]: Ext. s:3 @ from-trunk: Friendly Scanner from 10.226.31.1
  140.     -- Executing [s@from-trunk:4] Wait("SIP/SIP_Incoming-0000001b", "2") in new stack
  141. Retransmitting #2 (NAT) to 10.226.31.1:5060:
  142. SIP/2.0 200 OK
  143. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c;received=10.226.31.1;rport=5060
  144. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  145. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  146. To: <sip:4401@10.225.172.241>;tag=as6abe70aa
  147. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  148. CSeq: 1760593249 INVITE
  149. Server: FPBX-13.0.190.7(13.12.2)
  150. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  151. Supported: replaces, timer
  152. Session-Expires: 1800;refresher=uas
  153. Contact: <sip:4401@10.226.31.1:5060>
  154. Content-Type: application/sdp
  155. Require: timer
  156. Content-Length: 272
  157.  
  158. v=0
  159. o=root 720334440 720334440 IN IP4 10.226.31.1
  160. s=Asterisk PBX 13.12.2
  161. c=IN IP4 10.226.31.1
  162. t=0 0
  163. m=audio 15036 RTP/AVP 8 0 101
  164. a=rtpmap:8 PCMA/8000
  165. a=rtpmap:0 PCMU/8000
  166. a=rtpmap:101 telephone-event/8000
  167. a=fmtp:101 0-16
  168. a=ptime:20
  169. a=maxptime:150
  170. a=sendrecv
  171.  
  172. ---
  173.     -- Executing [s@from-trunk:5] Playback("SIP/SIP_Incoming-0000001b", "ss-noservice") in new stack
  174.     -- <SIP/SIP_Incoming-0000001b> Playing 'ss-noservice.alaw' (language 'en')
  175. Retransmitting #3 (NAT) to 10.226.31.1:5060:
  176. SIP/2.0 200 OK
  177. Via: SIP/2.0/UDP 10.226.31.1:5060;branch=z9hG4bK-*2*21687fbc09e05ac0c58c;received=10.226.31.1;rport=5060
  178. Record-Route: <sip:SBC_DIAG_2_0_02196e78@10.226.31.1:5060;lr>
  179. From: <sip:98475330@10.226.31.1>;tag=11e13ac-ExZa52b12a38124
  180. To: <sip:4401@10.225.172.241>;tag=as6abe70aa
  181. Call-ID: 587f800252b12a39-0292-2076@172.19.30.1
  182. CSeq: 1760593249 INVITE
  183. Server: FPBX-13.0.190.7(13.12.2)
  184. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  185. Supported: replaces, timer
  186. Session-Expires: 1800;refresher=uas
  187. Contact: <sip:4401@10.226.31.1:5060>
  188. Content-Type: application/sdp
  189. Require: timer
  190. Content-Length: 272
  191.  
  192. v=0
  193. o=root 720334440 720334440 IN IP4 10.226.31.1
  194. s=Asterisk PBX 13.12.2
  195. c=IN IP4 10.226.31.1
  196. t=0 0
  197. m=audio 15036 RTP/AVP 8 0 101
  198. a=rtpmap:8 PCMA/8000
  199. a=rtpmap:0 PCMU/8000
  200. a=rtpmap:101 telephone-event/8000
  201. a=fmtp:101 0-16
  202. a=ptime:20
  203. a=maxptime:150
  204. a=sendrecv

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