<div>Hi Duane,</div><div> </div><div>Thanks for your answer.</div><div> </div><div>By internal calls I mean that 2 numbers in the Customers list are calling eachother and being routed back to the same IP-address.</div><div>
 </div><div>Asterisk by default sends all calls to CDRTool, but when the call is routed back because it is a internal customer, then CDRTool does not even log or bill the call. Freeradius logs: Acct-Status-Type = Failed</div>
<div> </div><div>We have tried to look into your suggestions but it did not help.</div><div>Can it be Asterisk reinvite issue?</div><div> </div><div>Best regards,<br>Tony<br><br></div><div class="gmail_quote">2011/6/21  <span dir="ltr">&lt;<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>&gt;</span><br>
<blockquote style="margin: 0px 0px 0px 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">Look at my last post here
<br>
<br><a href="http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-CDR-Flow-record-td6432731.html#a6449882" target="_blank">http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-CDR-Flow-record-td6432731.html#a6449882</a>
<br>
<br>I think it&#39;s just a matter of what you have configured for
<br>&quot;E164_class&quot;         =&gt; 
<br>&quot;intAccessCode&quot;      =&gt; 
<br>&quot;natAccessCode&quot;      =&gt; 
<br>
<br>And also edit the can_uri.  So if your can_uri for your internal calls is set up the same as outbound calls it should calculate the calls.
<br>
<br>Hope that answers the issue you are having.
<br><div class="im">
<br>
<br>
<br>On Jun 21, 2011 7:29am, Tony Tyler &lt;<a href="mailto:tonytyler2011@gmail.com" target="_blank">tonytyler2011@gmail.com</a>&gt; wrote:
<br>&gt; Anyone?
<br>&gt;  
<br>&gt; From FreeRadius log:
<br>&gt;  
<br>&gt; Acct-Status-Type = Failed
<br>&gt;         Service-Type = Sip-Session
<br>&gt;         Sip-Response-Code = 408
<br>&gt;         Sip-Method = Invite
<br>&gt; 
<br>&gt; 
<br>&gt;         Event-Timestamp = &quot;Jun 14 2011 16:02:47 CEST&quot;
<br>&gt;         Sip-From-Tag = &quot;as16538a8e&quot;
<br>&gt;         Acct-Session-Id = &quot;18375a0e058480f5626c94293975f7cf@x&quot;
<br>&gt; 
<br>&gt; 
<br>&gt;         User-Name = &quot;a@x&quot;
<br>&gt;         Calling-Station-Id = &quot;sip:a@x&quot;
<br>&gt;         Called-Station-Id = &quot;sip:b@y&quot;
<br>&gt;         Sip-Translated-Request-URI = &quot;sip:b@ip&quot;
<br>&gt; 
<br>&gt; 
<br>&gt;         Source-IP = &quot;ip&quot;
<br>&gt;         Source-Port = &quot;5060&quot;
<br>&gt;         Billing-Party = &quot;sip:a@x&quot;
<br>&gt;         Canonical-URI = &quot;sip:b@y&quot;
<br>&gt;         User-Agent = &quot;Asterisk PBX&quot;
<br>&gt; 
<br>&gt; 
<br>&gt;         Contact = &quot;&quot;
<br></div><div><div></div><div class="h5">&gt;         NAS-Port = 5060
<br>&gt;         Acct-Delay-Time = 0
<br>&gt;         NAS-IP-Address = 127.0.0.1
<br>&gt;         Acct-Unique-Session-Id = &quot;89f0c9014ba0a2f9&quot;
<br>&gt;         Timestamp = 1308060167
<br>&gt; 
<br>&gt; 
<br>&gt;         Request-Authenticator = Verified
<br>&gt; 
<br>&gt;   Best regards,
<br>&gt; Tony Tyler
<br>&gt; 
<br>&gt; 
<br>&gt; 2011/6/14 Tony Tyler <a href="mailto:tonytyler2011@gmail.com" target="_blank">tonytyler2011@gmail.com</a>&gt;
<br>&gt; 
<br>&gt; 
<br>&gt; Hi,
<br>&gt;  
<br>&gt; We have setup CDRTool version 8.0.15 with an Asterisk multitenant PBX.
<br>&gt; 
<br>&gt; 
<br>&gt;  
<br>&gt; We want to be able to log and bill the internal calls in CDRTool.  
<br>&gt; All the calls are sent from Asterisk to OpenSIPS and the call is sent back to the same Asterisk on the same IP-address.
<br>&gt; 
<br>&gt; 
<br>&gt;  
<br>&gt; If it´s a external call, there is no problem. The problem occurs when the called number is a local customer, then it can´t log or bill the call in CDRTool.
<br>&gt;  
<br>&gt; From FreeRadius log:
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt;  
<br>&gt;  Acct-Status-Type = Failed
<br>&gt;         Service-Type = Sip-Session
<br>&gt;         Sip-Response-Code = 408
<br>&gt;         Sip-Method = Invite
<br>&gt;         Event-Timestamp = &quot;Jun 14 2011 16:02:47 CEST&quot;
<br>&gt;         Sip-From-Tag = &quot;as16538a8e&quot;
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt;         Acct-Session-Id = &quot;18375a0e058480f5626c94293975f7cf@x&quot;
<br>&gt;         User-Name = &quot;a@x&quot;
<br>&gt; 
<br>&gt; 
<br>&gt;         Calling-Station-Id = &quot;sip:a@x&quot;
<br>&gt; 
<br>&gt; 
<br>&gt;         Called-Station-Id = &quot;sip:b@y&quot;
<br>&gt;         Sip-Translated-Request-URI = &quot;sip:b@ip&quot;
<br>&gt;         Source-IP = &quot;ip&quot;
<br>&gt;         Source-Port = &quot;5060&quot;
<br>&gt;         Billing-Party = &quot;sip:a@x&quot;
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt;         Canonical-URI = &quot;sip:b@y&quot;
<br>&gt;         User-Agent = &quot;Asterisk PBX&quot;
<br>&gt;         Contact = &quot;&quot;
<br>&gt;         NAS-Port = 5060
<br>&gt;         Acct-Delay-Time = 0
<br>&gt;         NAS-IP-Address = 127.0.0.1
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt;         Acct-Unique-Session-Id = &quot;89f0c9014ba0a2f9&quot;
<br>&gt;         Timestamp = 1308060167
<br>&gt;         Request-Authenticator = Verified
<br>&gt;  
<br>&gt;  
<br>&gt; Best regards,
<br>&gt; Tony Tyler
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt; 
<br>&gt;</div></div><br>_______________________________________________<br>
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<br></blockquote></div><br>