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  1. --- onNewCallCreated b7504400: ooh323c_11
  2. +++ onNewCallCreated ooh323c_11
  3. --- ooh323_onReceivedSetup ooh323c_11
  4. --- ooh323_alloc
  5. +++ ooh323_alloc
  6. --- find_user: 9001112233 [mypbx], 10.20.182.78
  7. +++ find_user
  8. Adding capabilities to call(incoming, ooh323c_11)
  9. Adding g729A capability to call(incoming, ooh323c_11)
  10. Adding g729 capability to call(incoming, ooh323c_11)
  11. Adding g729B capability to call(incoming, ooh323c_11)
  12. --- configure_local_rtp
  13. +++ configure_local_rtp
  14. --- ooh323_new - uplink-h323, 0
  15. +++ h323_new
  16. +++ ooh323_onReceivedSetup - Determined context from_uplink, extension 1224499
  17. -- Executing [1224499@from_uplink:1] Answer("OOH323/uplink-h323-10", "") in new stack
  18. --- ooh323_answer
  19. +++ ooh323_answer
  20. ----- ooh323_indicate -1 on call ooh323c_11
  21. ++++ ooh323_indicate -1 on ooh323c_11
  22. -- Executing [1224499@from_uplink:2] Dial("OOH323/uplink-h323-10", "SIP/testuser") in new stack
  23. == Using SIP RTP CoS mark 5
  24. Audio is at 15646
  25. Adding codec 0x100 (g729) to SDP
  26. Adding non-codec 0x1 (telephone-event) to SDP
  27. Reliably Transmitting (NAT) to 90011122336.2:55726:
  28. INVITE sip:testuser@172.16.0.137:5060 SIP/2.0
  29. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  30. Max-Forwards: 70
  31. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  32. To: <sip:testuser@172.16.0.137:5060>
  33. Contact: <sip:9001112233@10.20.182.103:5060>
  34. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  35. CSeq: 102 INVITE
  36. User-Agent: Asterisk PBX 1.8.15.1
  37. Date: Tue, 23 Oct 2012 17:38:26 GMT
  38. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  39. Supported: replaces, timer
  40. Content-Type: application/sdp
  41. Content-Length: 291
  42.  
  43. v=0
  44. o=root 1323698518 1323698518 IN IP4 10.20.182.103
  45. s=Asterisk PBX 1.8.15.1
  46. c=IN IP4 10.20.182.103
  47. t=0 0
  48. m=audio 15646 RTP/AVP 18 101
  49. a=rtpmap:18 G729/8000
  50. a=fmtp:18 annexb=no
  51. a=rtpmap:101 telephone-event/8000
  52. a=fmtp:101 0-16
  53. a=silenceSupp:off - - - -
  54. a=ptime:20
  55. a=sendrecv
  56.  
  57. ---
  58. -- Called SIP/testuser
  59. ----- ooh323_indicate 22 on call ooh323c_11
  60. ++++ ooh323_indicate 22 on ooh323c_11
  61. ----- ooh323_indicate 22 on call ooh323c_11
  62. ++++ ooh323_indicate 22 on ooh323c_11
  63.  
  64. <--- SIP read from UDP:90011122336.2:55726 --->
  65. SIP/2.0 100 Trying
  66. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  67. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  68. To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
  69. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  70. CSeq: 102 INVITE
  71. Supported: em,timer,replaces,path,early-session,resource-priority
  72. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
  73. Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
  74. Content-Length: 0
  75.  
  76. <------------->
  77. --- (10 headers 0 lines) ---
  78.  
  79. <--- SIP read from UDP:90011122336.2:55726 --->
  80. SIP/2.0 183 Session Progress
  81. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  82. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  83. To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
  84. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  85. CSeq: 102 INVITE
  86. Contact: <sip:testuser@172.16.0.137:5060>
  87. Supported: em,timer,replaces,path,early-session,resource-priority
  88. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
  89. Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
  90. Content-Type: application/sdp
  91. Content-Length: 256
  92.  
  93. v=0
  94. o=AudiocodesGW 2029396522 2029396385 IN IP4 172.16.0.137
  95. s=Phone-Call
  96. c=IN IP4 172.16.0.137
  97. t=0 0
  98. m=audio 6030 RTP/AVP 18 101
  99. a=rtpmap:18 G729/8000
  100. a=fmtp:18 annexb=no
  101. a=rtpmap:101 telephone-event/8000
  102. a=fmtp:101 0-15
  103. a=ptime:20
  104. a=sendrecv
  105. <------------->
  106. --- (12 headers 12 lines) ---
  107. list_route: hop: <sip:testuser@172.16.0.137:5060>
  108. Found RTP audio format 18
  109. Found RTP audio format 101
  110. Found audio description format G729 for ID 18
  111. Found audio description format telephone-event for ID 101
  112. Capabilities: us - 0x80000008010e (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  113. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  114. Peer audio RTP is at port 172.16.0.137:6030
  115. -- SIP/testuser-0000000a is making progress passing it to OOH323/uplink-h323-10
  116. ----- ooh323_indicate 14 on call ooh323c_11
  117. ++++ ooh323_indicate 14 on ooh323c_11
  118. --- onCallEstablished ooh323c_11
  119. --- find_call
  120. +++ find_call
  121. +++ onCallEstablished ooh323c_11
  122. --- onCallCleared ooh323c_11
  123. --- find_call
  124. +++ find_call
  125. +++ onCallCleared
  126. Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)
  127. Reliably Transmitting (NAT) to 90011122336.2:55726:
  128. CANCEL sip:testuser@172.16.0.137:5060 SIP/2.0
  129. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  130. Max-Forwards: 70
  131. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  132. To: <sip:testuser@172.16.0.137:5060>
  133. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  134. CSeq: 102 CANCEL
  135. User-Agent: Asterisk PBX 1.8.15.1
  136. Content-Length: 0
  137.  
  138.  
  139. ---
  140. Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)
  141. == Spawn extension (from_uplink, 1224499, 2) exited non-zero on 'OOH323/uplink-h323-10'
  142. --- ooh323_hangup
  143. +++ ooh323_hangup
  144.  
  145. <--- SIP read from UDP:90011122336.2:55726 --->
  146. SIP/2.0 487 Request Terminated
  147. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  148. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  149. To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
  150. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  151. CSeq: 102 INVITE
  152. Supported: em,timer,replaces,path,early-session,resource-priority
  153. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
  154. Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
  155. Reason: SIP ;cause=487 ;text="487 Request Terminated"
  156. Content-Length: 0
  157.  
  158. <------------->
  159. --- (11 headers 0 lines) ---
  160. Transmitting (NAT) to 90011122336.2:55726:
  161. ACK sip:testuser@172.16.0.137:5060 SIP/2.0
  162. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  163. Max-Forwards: 70
  164. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  165. To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
  166. Contact: <sip:9001112233@10.20.182.103:5060>
  167. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  168. CSeq: 102 ACK
  169. User-Agent: Asterisk PBX 1.8.15.1
  170. Content-Length: 0
  171.  
  172.  
  173. ---
  174. Scheduling destruction of SIP dialog '7af4263648279b276678d03372793143@10.20.182.103:5060' in 32000 ms (Method: INVITE)
  175.  
  176. <--- SIP read from UDP:90011122336.2:55726 --->
  177. SIP/2.0 200 OK
  178. Via: SIP/2.0/UDP 10.20.182.103:5060;branch=z9hG4bK714357be;rport
  179. From: "9001112233 [mypbx]" <sip:9001112233@10.20.182.103>;tag=as0e6552ec
  180. To: <sip:testuser@172.16.0.137:5060>;tag=1c2029375717
  181. Call-ID: 7af4263648279b276678d03372793143@10.20.182.103:5060
  182. CSeq: 102 CANCEL
  183. Contact: <sip:testuser@172.16.0.137:5060>
  184. Supported: em,timer,replaces,path,early-session,resource-priority
  185. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
  186. Server: Audiocodes-Sip-Gateway-MP-114 FXS/v.6.40A.011.008
  187. Content-Length: 0
  188.  
  189. <------------->
  190. --- (11 headers 0 lines) ---
  191. --- ooh323_destroy
  192. Destroying uplink-h323
  193. Destroying ooh323c_11
  194. --- find_user: (null), 10.20.182.78
  195. +++ find_user
  196. +++ ooh323_destroy
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