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  1. <--- SIP read from UDP:209.87.247.153:5060 --->
  2. INVITE sip:[email protected];user=phone SIP/2.0
  3. Record-Route: <sip:209.87.247.153;lr=on;ftag=CF008F69-5B5EA8BE;nat=yes>
  4. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKbdb2.8f1df851.0
  5. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bKcfaea0c87B7A97E1
  6. From: "Paul Belanger" <sip:[email protected]>;tag=CF008F69-5B5EA8BE
  7. To: <sip:[email protected];user=phone>
  8. CSeq: 2 INVITE
  9. Call-ID: [email protected]
  10. Contact: <sip:[email protected];alias=24.52.251.44~1024~1>
  11. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  12. User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
  13. Accept-Language: en
  14. Supported: 100rel,replaces
  15. Allow-Events: talk,hold,conference
  16. Max-Forwards: 69
  17. Content-Type: application/sdp
  18. Content-Length: 317
  19.  
  20. v=0
  21. o=- 1167610650 1167610650 IN IP4 209.87.247.153
  22. s=Polycom IP Phone
  23. c=IN IP4 209.87.247.153
  24. t=0 0
  25. a=sendrecv
  26. m=audio 42142 RTP/AVP 9 0 8 18 127
  27. a=rtpmap:9 G722/8000
  28. a=rtpmap:0 PCMU/8000
  29. a=rtpmap:8 PCMA/8000
  30. a=rtpmap:18 G729/8000
  31. a=fmtp:18 annexb=no
  32. a=rtpmap:127 telephone-event/8000
  33. a=nortpproxy:yes
  34. <------------->
  35. --- (17 headers 14 lines) ---
  36. Sending to 209.87.247.153:5060 (no NAT)
  37. Using INVITE request as basis request - [email protected]
  38. Found peer 'kamailio-01-prod' for '213' from 209.87.247.153:5060
  39. == Using SIP RTP CoS mark 5
  40. Found RTP audio format 9
  41. Found RTP audio format 0
  42. Found RTP audio format 8
  43. Found RTP audio format 18
  44. Found RTP audio format 127
  45. Found audio description format G722 for ID 9
  46. Found audio description format PCMU for ID 0
  47. Found audio description format PCMA for ID 8
  48. Found audio description format G729 for ID 18
  49. Found audio description format telephone-event for ID 127
  50. Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  51. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  52. Peer audio RTP is at port 209.87.247.153:42142
  53. Looking for 6134545038 in from-kamailio (domain polybeacon.com)
  54. list_route: hop: <sip:209.87.247.153;lr=on;ftag=CF008F69-5B5EA8BE;nat=yes>
  55.  
  56. <--- Transmitting (no NAT) to 209.87.247.153:5060 --->
  57. SIP/2.0 100 Trying
  58. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKbdb2.8f1df851.0;received=209.87.247.153
  59. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bKcfaea0c87B7A97E1
  60. Record-Route: <sip:209.87.247.153;lr=on;ftag=CF008F69-5B5EA8BE;nat=yes>
  61. From: "Paul Belanger" <sip:[email protected]>;tag=CF008F69-5B5EA8BE
  62. To: <sip:[email protected];user=phone>
  63. Call-ID: [email protected]
  64. CSeq: 2 INVITE
  65. Server: Asterisk PBX 1.8.7.1-1kickstand0~ppa189.1~precise
  66. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  67. Supported: replaces, timer
  68. Contact: <sip:[email protected]:15060>
  69. Content-Length: 0
  70.  
  71.  
  72. <------------>
  73. -- Executing [6134545038@from-kamailio:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  74. -- Executing [6134545038@from-kamailio:2] NoOp("SIP/kamailio-01-prod-00000002", "polybeacon.com") in new stack
  75. -- Executing [6134545038@from-kamailio:3] GotoIf("SIP/kamailio-01-prod-00000002", "1?polybeacon.com,6134545038,1") in new stack
  76. -- Goto (polybeacon.com,6134545038,1)
  77. -- Executing [[email protected]:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  78. -- Executing [[email protected]:2] Gosub("SIP/kamailio-01-prod-00000002", "IVR-GoToContext,s,1(ivr-polybeacon.com-start,s,1)") in new stack
  79. -- Executing [s@IVR-GoToContext:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  80. -- Executing [s@IVR-GoToContext:2] Set("SIP/kamailio-01-prod-00000002", "INVALID_COUNT=0") in new stack
  81. -- Executing [s@IVR-GoToContext:3] Set("SIP/kamailio-01-prod-00000002", "TIMEOUT_COUNT=0") in new stack
  82. -- Executing [s@IVR-GoToContext:4] Wait("SIP/kamailio-01-prod-00000002", ".5") in new stack
  83. -- Executing [s@IVR-GoToContext:5] Gosub("SIP/kamailio-01-prod-00000002", "ivr-polybeacon.com-start,s,1") in new stack
  84. -- Executing [[email protected]:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  85. -- Executing [[email protected]:2] Answer("SIP/kamailio-01-prod-00000002", "") in new stack
  86. Audio is at 15060
  87. Adding codec 0x4 (ulaw) to SDP
  88. Adding codec 0x8 (alaw) to SDP
  89. Adding non-codec 0x1 (telephone-event) to SDP
  90.  
  91. <--- Reliably Transmitting (no NAT) to 209.87.247.153:5060 --->
  92. SIP/2.0 200 OK
  93. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKbdb2.8f1df851.0;received=209.87.247.153
  94. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bKcfaea0c87B7A97E1
  95. Record-Route: <sip:209.87.247.153;lr=on;ftag=CF008F69-5B5EA8BE;nat=yes>
  96. From: "Paul Belanger" <sip:[email protected]>;tag=CF008F69-5B5EA8BE
  97. To: <sip:[email protected];user=phone>;tag=as3d681da7
  98. Call-ID: [email protected]
  99. CSeq: 2 INVITE
  100. Server: Asterisk PBX 1.8.7.1-1kickstand0~ppa189.1~precise
  101. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  102. Supported: replaces, timer
  103. Contact: <sip:[email protected]:15060>
  104. Content-Type: application/sdp
  105. Content-Length: 318
  106.  
  107. v=0
  108. o=root 614056137 614056137 IN IP4 209.87.247.153
  109. s=Asterisk PBX 1.8.7.1-1kickstand0~ppa189.1~precise
  110. c=IN IP4 209.87.247.153
  111. t=0 0
  112. m=audio 11642 RTP/AVP 0 8 127
  113. a=rtpmap:0 PCMU/8000
  114. a=rtpmap:8 PCMA/8000
  115. a=rtpmap:127 telephone-event/8000
  116. a=fmtp:127 0-16
  117. a=silenceSupp:off - - - -
  118. a=ptime:20
  119. a=sendrecv
  120.  
  121. <------------>
  122.  
  123. <--- SIP read from UDP:209.87.247.153:5060 --->
  124. ACK sip:[email protected]:15060 SIP/2.0
  125. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKcydzigwkX
  126. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK6576c75458DBD3D
  127. From: "Paul Belanger" <sip:[email protected]>;tag=CF008F69-5B5EA8BE
  128. To: <sip:[email protected];user=phone>;tag=as3d681da7
  129. CSeq: 2 ACK
  130. Call-ID: [email protected]
  131. Contact: <sip:[email protected];alias=24.52.251.44~1024~1>
  132. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  133. User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
  134. Accept-Language: en
  135. Max-Forwards: 69
  136. Content-Length: 0
  137.  
  138. <------------->
  139. --- (13 headers 0 lines) ---
  140. -- Executing [[email protected]:3] Set("SIP/kamailio-01-prod-00000002", "IVR_DOMAIN=polybeacon.com") in new stack
  141. -- Executing [[email protected]:4] Set("SIP/kamailio-01-prod-00000002", "TIMEOUT(digits)=3") in new stack
  142. -- Digit timeout set to 3.000
  143. -- Executing [[email protected]:5] Gosub("SIP/kamailio-01-prod-00000002", "IVR-GoToContext,s,1(ivr-polybeacon.com,s,1)") in new stack
  144. -- Executing [s@IVR-GoToContext:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  145. -- Executing [s@IVR-GoToContext:2] Set("SIP/kamailio-01-prod-00000002", "INVALID_COUNT=0") in new stack
  146. -- Executing [s@IVR-GoToContext:3] Set("SIP/kamailio-01-prod-00000002", "TIMEOUT_COUNT=0") in new stack
  147. -- Executing [s@IVR-GoToContext:4] Wait("SIP/kamailio-01-prod-00000002", ".5") in new stack
  148. -- Executing [s@IVR-GoToContext:5] Gosub("SIP/kamailio-01-prod-00000002", "ivr-polybeacon.com,s,1") in new stack
  149. -- Executing [[email protected]:1] NoOp("SIP/kamailio-01-prod-00000002", "") in new stack
  150. -- Executing [[email protected]:2] BackGround("SIP/kamailio-01-prod-00000002", "custom/polybeacon.com/1") in new stack
  151. -- <SIP/kamailio-01-prod-00000002> Playing 'custom/polybeacon.com/1.gsm' (language 'en')
  152.  
  153. <--- SIP read from UDP:209.87.247.153:5060 --->
  154. BYE sip:[email protected]:15060 SIP/2.0
  155. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKcdb2.ae0627d4.0
  156. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK33404e42E6AB52F3
  157. From: "Paul Belanger" <sip:[email protected]>;tag=CF008F69-5B5EA8BE
  158. To: <sip:[email protected];user=phone>;tag=as3d681da7
  159. CSeq: 3 BYE
  160. Call-ID: [email protected]
  161. Contact: <sip:[email protected];alias=24.52.251.44~1024~1>
  162. User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
  163. Accept-Language: en
  164. Proxy-Authorization: Digest username="[email protected]", realm="polybeacon.com", nonce="UjSu+VI0rc0bvVfdBDx9FwlBb8FdjKZ5", uri="sip:[email protected];user=phone", response="f9a38f5930249e5cbc32679f5a689f77", algorithm=MD5
  165. Max-Forwards: 69
  166. Content-Length: 0
  167.  
  168. <------------->
  169. pbx-01-prod*CLI> sip show peers
  170. Name/username Host Dyn Forcerport ACL Port Status Description
  171. kamailio 209.87.247.153 5060 OK (1 ms)
  172. kamailio-01-prod 209.87.247.153 5060 OK (1 ms)
  173. voip.ms 184.75.215.106 5060 OK (8 ms)
  174. 3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline]
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