System B PJSIP Log

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  1. <--- Received SIP request (925 bytes) from UDP:172.16.1.12:5060 --->
  2. INVITE sip:4385551234@172.16.1.10:5060 SIP/2.0
  3. Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  4. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  5. To: <sip:4385551234@172.16.1.10>
  6. Contact: <sip:asterisk@104.145.12.182:5060>
  7. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  8. CSeq: 6113 INVITE
  9. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  10. Supported: 100rel, timer, replaces, norefersub
  11. Session-Expires: 1800
  12. Min-SE: 90
  13. Max-Forwards: 70
  14. User-Agent: FreePBX-15.0.16.53(16.9.0)
  15. Content-Type: application/sdp
  16. Content-Length:   239
  17.  
  18. v=0
  19. o=- 373719029 373719029 IN IP4 104.145.12.182
  20. s=Asterisk
  21. c=IN IP4 104.145.12.182
  22. t=0 0
  23. m=audio 16050 RTP/AVP 0 101
  24. a=rtpmap:0 PCMU/8000
  25. a=rtpmap:101 telephone-event/8000
  26. a=fmtp:101 0-16
  27. a=ptime:20
  28. a=maxptime:150
  29. a=sendrecv
  30.  
  31. <--- Transmitting SIP response (389 bytes) to UDP:172.16.1.12:5060 --->
  32. SIP/2.0 100 Trying
  33. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  34. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  35. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  36. To: <sip:4385551234@172.16.1.10>
  37. CSeq: 6113 INVITE
  38. Server: FreePBX-15.0.16.53(16.9.0)
  39. Content-Length:  0
  40.  
  41.  
  42. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  43. SIP/2.0 200 OK
  44. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  45. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  46. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  47. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  48. CSeq: 6113 INVITE
  49. Server: FreePBX-15.0.16.53(16.9.0)
  50. Contact: <sip:104.145.12.182:5060>
  51. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  52. Supported: 100rel, timer, replaces, norefersub
  53. Session-Expires: 1800;refresher=uac
  54. Require: timer
  55. Content-Type: application/sdp
  56. Content-Length:   239
  57.  
  58. v=0
  59. o=- 373719029 373719031 IN IP4 104.145.12.182
  60. s=Asterisk
  61. c=IN IP4 104.145.12.182
  62. t=0 0
  63. m=audio 13080 RTP/AVP 0 101
  64. a=rtpmap:0 PCMU/8000
  65. a=rtpmap:101 telephone-event/8000
  66. a=fmtp:101 0-16
  67. a=ptime:20
  68. a=maxptime:150
  69. a=sendrecv
  70.  
  71. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  72. SIP/2.0 200 OK
  73. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  74. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  75. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  76. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  77. CSeq: 6113 INVITE
  78. Server: FreePBX-15.0.16.53(16.9.0)
  79. Contact: <sip:104.145.12.182:5060>
  80. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  81. Supported: 100rel, timer, replaces, norefersub
  82. Session-Expires: 1800;refresher=uac
  83. Require: timer
  84. Content-Type: application/sdp
  85. Content-Length:   239
  86.  
  87. v=0
  88. o=- 373719029 373719031 IN IP4 104.145.12.182
  89. s=Asterisk
  90. c=IN IP4 104.145.12.182
  91. t=0 0
  92. m=audio 13080 RTP/AVP 0 101
  93. a=rtpmap:0 PCMU/8000
  94. a=rtpmap:101 telephone-event/8000
  95. a=fmtp:101 0-16
  96. a=ptime:20
  97. a=maxptime:150
  98. a=sendrecv
  99.  
  100. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  101. SIP/2.0 200 OK
  102. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  103. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  104. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  105. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  106. CSeq: 6113 INVITE
  107. Server: FreePBX-15.0.16.53(16.9.0)
  108. Contact: <sip:104.145.12.182:5060>
  109. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  110. Supported: 100rel, timer, replaces, norefersub
  111. Session-Expires: 1800;refresher=uac
  112. Require: timer
  113. Content-Type: application/sdp
  114. Content-Length:   239
  115.  
  116. v=0
  117. o=- 373719029 373719031 IN IP4 104.145.12.182
  118. s=Asterisk
  119. c=IN IP4 104.145.12.182
  120. t=0 0
  121. m=audio 13080 RTP/AVP 0 101
  122. a=rtpmap:0 PCMU/8000
  123. a=rtpmap:101 telephone-event/8000
  124. a=fmtp:101 0-16
  125. a=ptime:20
  126. a=maxptime:150
  127. a=sendrecv
  128.  
  129. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  130. SIP/2.0 200 OK
  131. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  132. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  133. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  134. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  135. CSeq: 6113 INVITE
  136. Server: FreePBX-15.0.16.53(16.9.0)
  137. Contact: <sip:104.145.12.182:5060>
  138. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  139. Supported: 100rel, timer, replaces, norefersub
  140. Session-Expires: 1800;refresher=uac
  141. Require: timer
  142. Content-Type: application/sdp
  143. Content-Length:   239
  144.  
  145. v=0
  146. o=- 373719029 373719031 IN IP4 104.145.12.182
  147. s=Asterisk
  148. c=IN IP4 104.145.12.182
  149. t=0 0
  150. m=audio 13080 RTP/AVP 0 101
  151. a=rtpmap:0 PCMU/8000
  152. a=rtpmap:101 telephone-event/8000
  153. a=fmtp:101 0-16
  154. a=ptime:20
  155. a=maxptime:150
  156. a=sendrecv
  157.  
  158. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  159. SIP/2.0 200 OK
  160. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  161. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  162. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  163. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  164. CSeq: 6113 INVITE
  165. Server: FreePBX-15.0.16.53(16.9.0)
  166. Contact: <sip:104.145.12.182:5060>
  167. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  168. Supported: 100rel, timer, replaces, norefersub
  169. Session-Expires: 1800;refresher=uac
  170. Require: timer
  171. Content-Type: application/sdp
  172. Content-Length:   239
  173.  
  174. v=0
  175. o=- 373719029 373719031 IN IP4 104.145.12.182
  176. s=Asterisk
  177. c=IN IP4 104.145.12.182
  178. t=0 0
  179. m=audio 13080 RTP/AVP 0 101
  180. a=rtpmap:0 PCMU/8000
  181. a=rtpmap:101 telephone-event/8000
  182. a=fmtp:101 0-16
  183. a=ptime:20
  184. a=maxptime:150
  185. a=sendrecv
  186.  
  187. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  188. SIP/2.0 200 OK
  189. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  190. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  191. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  192. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  193. CSeq: 6113 INVITE
  194. Server: FreePBX-15.0.16.53(16.9.0)
  195. Contact: <sip:104.145.12.182:5060>
  196. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  197. Supported: 100rel, timer, replaces, norefersub
  198. Session-Expires: 1800;refresher=uac
  199. Require: timer
  200. Content-Type: application/sdp
  201. Content-Length:   239
  202.  
  203. v=0
  204. o=- 373719029 373719031 IN IP4 104.145.12.182
  205. s=Asterisk
  206. c=IN IP4 104.145.12.182
  207. t=0 0
  208. m=audio 13080 RTP/AVP 0 101
  209. a=rtpmap:0 PCMU/8000
  210. a=rtpmap:101 telephone-event/8000
  211. a=fmtp:101 0-16
  212. a=ptime:20
  213. a=maxptime:150
  214. a=sendrecv
  215.  
  216. <--- Transmitting SIP response (947 bytes) to UDP:172.16.1.12:5060 --->
  217. SIP/2.0 200 OK
  218. Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;received=172.16.1.12;branch=z9hG4bKPj126b464a-4744-49d1-a0e8-237d5e5627f1
  219. Call-ID: 4812e7a0-27f5-43ec-b6fb-6e866a8fa2eb
  220. From: "5145551234" <sip:5145551234@172.16.1.12>;tag=36c45c8f-535c-4375-8a8f-7ffbee551dd1
  221. To: <sip:4385551234@172.16.1.10>;tag=c59e4575-db40-4964-a16a-a5be1f57a9d7
  222. CSeq: 6113 INVITE
  223. Server: FreePBX-15.0.16.53(16.9.0)
  224. Contact: <sip:104.145.12.182:5060>
  225. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  226. Supported: 100rel, timer, replaces, norefersub
  227. Session-Expires: 1800;refresher=uac
  228. Require: timer
  229. Content-Type: application/sdp
  230. Content-Length:   239
  231.  
  232. v=0
  233. o=- 373719029 373719031 IN IP4 104.145.12.182
  234. s=Asterisk
  235. c=IN IP4 104.145.12.182
  236. t=0 0
  237. m=audio 13080 RTP/AVP 0 101
  238. a=rtpmap:0 PCMU/8000
  239. a=rtpmap:101 telephone-event/8000
  240. a=fmtp:101 0-16
  241. a=ptime:20
  242. a=maxptime:150
  243. a=sendrecv

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