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  1. == Using SIP RTP CoS mark 5
  2. Audio is at 5060
  3. Adding codec 0x100 (g729) to SDP
  4. Adding non-codec 0x1 (telephone-event) to SDP
  5. Reliably Transmitting (NAT) to 192.168.2.20:5080:
  6. INVITE sip:12212572422@192.168.2.20:5080 SIP/2.0
  7. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK4f4ed04c;rport
  8. Max-Forwards: 70
  9. From: "5142772007" <sip:5142772007@sweeny.test.com>;tag=as36cbb6e1
  10. To: <sip:12212572422@192.168.2.20:5080>
  11. Contact: <sip:5142772007@192.168.2.10:5060>
  12. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  13. CSeq: 102 INVITE
  14. User-Agent: SYMAX Systems
  15. Date: Tue, 10 Sep 2013 16:24:38 GMT
  16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  17. Supported: replaces, timer
  18. Content-Type: application/sdp
  19. Content-Length: 277
  20.  
  21. v=0
  22. o=root 666676249 666676249 IN IP4 192.168.2.10
  23. s=SYMAX Systems
  24. c=IN IP4 192.168.2.10
  25. t=0 0
  26. m=audio 14474 RTP/AVP 18 101
  27. a=rtpmap:18 G729/8000
  28. a=fmtp:18 annexb=no
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-16
  31. a=silenceSupp:off - - - -
  32. a=ptime:20
  33. a=sendrecv
  34.  
  35. ---
  36. -- Called SIP/VTrunk/12212572422
  37. [Sep 10 12:24:38] DEBUG[8220]: res_musiconhold.c:857 _get_mohbyname: Music on Hold class 'null' not found in memory
  38.  
  39. <--- SIP read from UDP:192.168.2.20:5080 --->
  40. SIP/2.0 100 Giving a try
  41. Via: SIP/2.0/UDP 192.168.2.10:5060;received=192.168.2.10;branch=z9hG4bK4f4ed04c;rport=5060
  42. From: "5142772007" <sip:5142772007@sweeny.test.com>;tag=as36cbb6e1
  43. To: <sip:12212572422@192.168.2.20:5080>
  44. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  45. CSeq: 102 INVITE
  46. Content-Length: 0
  47.  
  48. <------------->
  49. --- (7 headers 0 lines) ---
  50. [Sep 10 12:24:38] DEBUG[8220]: res_musiconhold.c:857 _get_mohbyname: Music on Hold class 'default' not found in memory
  51. -- Started music on hold, class 'default', on SIP/TTrunk-00000004
  52.  
  53. <--- SIP read from UDP:192.168.2.20:5080 --->
  54. SIP/2.0 183 Session Progress
  55. Via: SIP/2.0/UDP 192.168.2.10:5060;received=192.168.2.10;branch=z9hG4bK4f4ed04c;rport=5060
  56. Record-Route: <sip:21.21.29.23:5080;lr;did=0b8.c6b2d087>
  57. From: "5142772007" <sip:5142772007@sweeny.test.com>;tag=as36cbb6e1
  58. To: <sip:12212572422@192.168.2.20:5080>;tag=10.152.1.13+1+2362631+a4ec0644
  59. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  60. CSeq: 102 INVITE
  61. Server: DC-SIP/2.0
  62. Organization: MetaSwitch
  63. Contact: <sip:callee@108.59.2.133;did=0b8.cea47a65>
  64. Content-Length: 202
  65. Content-Type: application/sdp
  66. Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH
  67.  
  68. v=0
  69. o=- 145776966 145776967 IN IP4 69.95.249.148
  70. s=-
  71. c=IN IP4 192.168.2.20
  72. t=0 0
  73. m=audio 49660 RTP/AVP 18 101
  74. a=fmtp:18 annexb=no
  75. a=rtpmap:101 telephone-event/8000
  76. a=ptime:20
  77. a=nortpproxy:yes
  78. <------------->
  79. --- (13 headers 10 lines) ---
  80. Found RTP audio format 18
  81. Found RTP audio format 101
  82. Found audio description format telephone-event for ID 101
  83. Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  84. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  85. Peer audio RTP is at port 192.168.2.20:49660
  86. -- SIP/VTrunk-00000005 is making progress passing it to SIP/TTrunk-00000004
  87.  
  88. <--- SIP read from UDP:192.168.2.20:5080 --->
  89. SIP/2.0 200 OK
  90. Via: SIP/2.0/UDP 192.168.2.10:5060;received=192.168.2.10;branch=z9hG4bK4f4ed04c;rport=5060
  91. Record-Route: <sip:21.21.29.23:5080;lr;did=0b8.c6b2d087>
  92. From: "5142772007" <sip:5142772007@sweeny.test.com>;tag=as36cbb6e1
  93. To: <sip:12212572422@192.168.2.20:5080>;tag=10.152.1.13+1+2362631+a4ec0644
  94. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  95. CSeq: 102 INVITE
  96. Server: DC-SIP/2.0
  97. Organization: MetaSwitch
  98. Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
  99. Supported: resource-priority, 100rel
  100. Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH
  101. Accept-Encoding: identity
  102. Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml
  103. Contact: <sip:callee@108.59.2.133;did=0b8.cea47a65>
  104. Content-Length: 202
  105. Content-Type: application/sdp
  106.  
  107. v=0
  108. o=- 145776966 145776967 IN IP4 69.95.249.148
  109. s=-
  110. c=IN IP4 192.168.2.20
  111. t=0 0
  112. m=audio 49660 RTP/AVP 18 101
  113. a=fmtp:18 annexb=no
  114. a=rtpmap:101 telephone-event/8000
  115. a=ptime:20
  116. a=nortpproxy:yes
  117. <------------->
  118. --- (17 headers 10 lines) ---
  119. list_route: hop: <sip:21.21.29.23:5080;lr;did=0b8.c6b2d087>
  120. set_destination: Parsing <sip:21.21.29.23:5080;lr;did=0b8.c6b2d087> for address/port to send to
  121. set_destination: set destination to 21.21.29.23:5080
  122. Transmitting (NAT) to 192.168.2.20:5080:
  123. ACK sip:callee@108.59.2.133;did=0b8.cea47a65 SIP/2.0
  124. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK1900ed71;rport
  125. Route: <sip:21.21.29.23:5080;lr;did=0b8.c6b2d087>
  126. Max-Forwards: 70
  127. From: "5142772007" <sip:5142772007@sweeny.test.com>;tag=as36cbb6e1
  128. To: <sip:12212572422@192.168.2.20:5080>;tag=10.152.1.13+1+2362631+a4ec0644
  129. Contact: <sip:5142772007@192.168.2.10:5060>
  130. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  131. CSeq: 102 ACK
  132. User-Agent: SYMAX Systems
  133. Content-Length: 0
  134.  
  135.  
  136. ---
  137. -- SIP/VTrunk-00000005 answered SIP/TTrunk-00000004
  138. -- Stopped music on hold on SIP/TTrunk-00000004
  139.  
  140. <--- SIP read from UDP:192.168.2.20:5080 --->
  141. BYE sip:5142772007@192.168.2.10:5060 SIP/2.0
  142. Via: SIP/2.0/UDP 192.168.2.20:5080;branch=z9hG4bK0bc7.8269b7c2.0
  143. Via: SIP/2.0/UDP 108.59.2.133;rport=5060;received=108.59.2.133;branch=z9hG4bK0bc7.8d514e31.0
  144. Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
  145. Max-Forwards: 63
  146. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  147. From: <sip:12212572422@192.168.2.20:5080>;tag=10.152.1.13+1+2362631+a4ec0644
  148. To: "5142772007" <sip:5142772007@108.59.2.131>;tag=as36cbb6e1
  149. CSeq: 1020595767 BYE
  150. Organization: MetaSwitch
  151. Supported: resource-priority, 100rel
  152. Content-Length: 0
  153.  
  154. <------------->
  155. --- (12 headers 0 lines) ---
  156. Sending to 192.168.2.20:5080 (NAT)
  157. Scheduling destruction of SIP dialog '7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com' in 32000 ms (Method: BYE)
  158.  
  159. <--- Transmitting (NAT) to 192.168.2.20:5080 --->
  160. SIP/2.0 200 OK
  161. Via: SIP/2.0/UDP 192.168.2.20:5080;branch=z9hG4bK0bc7.8269b7c2.0;received=192.168.2.20;rport=5080
  162. Via: SIP/2.0/UDP 108.59.2.133;rport=5060;received=108.59.2.133;branch=z9hG4bK0bc7.8d514e31.0
  163. From: <sip:12212572422@192.168.2.20:5080>;tag=10.152.1.13+1+2362631+a4ec0644
  164. To: "5142772007" <sip:5142772007@108.59.2.131>;tag=as36cbb6e1
  165. Call-ID: 7bc6ecc70e5333d3314e0c9937614bff@sweeny.test.com
  166. CSeq: 1020595767 BYE
  167. Server: SYMAX Systems
  168. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  169. Supported: replaces, timer
  170. Content-Length: 0
  171.  
  172.  
  173. <------------>
  174. -- <SIP/TTrunk-00000004> Playing 'prepaid-enter-dest.gsm' (language 'en')
  175.  
  176. <--- SIP read from UDP:192.168.2.5:5060 --->
  177. BYE sip:6474770990@192.168.2.10:5060 SIP/2.0
  178. Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK18ca.0bca325.0
  179. Via: SIP/2.0/UDP 205.205.22.128;received=205.205.22.128;rport=5060;branch=z9hG4bKvHrNr22Hvm80a
  180. Max-Forwards: 69
  181. From: "NICK KHAMIS" <sip:5142772007@205.205.22.128>;tag=FZ6r9reKHZjBS
  182. To: <sip:6474770990@192.168.2.5:5060>;tag=as570e5222
  183. Call-ID: 32ba0f09-94d8-1231-19be-005056a149a4
  184. CSeq: 49079047 BYE
  185. Contact: <sip:gw+cust_sbc_trial_gw1@205.205.22.128:5060;transport=udp;gw=cust_sbc_trial_gw1>
  186. User-Agent: ISPT_b2bua
  187. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
  188. Supported: timer, precondition, path, replaces
  189. Reason: Q.850;cause=16;text="Normal clearing"
  190. Content-Length: 0
  191.  
  192. <------------->
  193. --- (14 headers 0 lines) ---
  194. Sending to 192.168.2.5:5060 (NAT)
  195. Scheduling destruction of SIP dialog '32ba0f09-94d8-1231-19be-005056a149a4' in 32000 ms (Method: BYE)
  196.  
  197. <--- Transmitting (NAT) to 192.168.2.5:5060 --->
  198. SIP/2.0 200 OK
  199. Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK18ca.0bca325.0;received=192.168.2.5;rport=5060
  200. Via: SIP/2.0/UDP 205.205.22.128;received=205.205.22.128;rport=5060;branch=z9hG4bKvHrNr22Hvm80a
  201. From: "NICK KHAMIS" <sip:5142772007@205.205.22.128>;tag=FZ6r9reKHZjBS
  202. To: <sip:6474770990@192.168.2.5:5060>;tag=as570e5222
  203. Call-ID: 32ba0f09-94d8-1231-19be-005056a149a4
  204. CSeq: 49079047 BYE
  205. Server: SYMAX Systems
  206. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  207. Supported: replaces, timer
  208. Content-Length: 0
  209.  
  210.  
  211. <------------>
  212. -- <SIP/TTrunk-00000004>AGI Script a2billing.php completed, returning 4
  213. == Spawn extension (context-from-peere, 6474770990, 2) exited non-zero on 'SIP/TTrunk-00000004'
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