[OpenSIPS-Users] CDRTool billing of internal calls

[duane.larson at gmail.com]

Look at my last post here

http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-CDR-Flow-record-td6432731.html#a6449882

I think it's just a matter of what you have configured for
"E164_class" =>
"intAccessCode" =>
"natAccessCode" =>

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.

Hope that answers the issue you are having.



On Jun 21, 2011 7:29am, Tony Tyler <tonytyler2011 at gmail.com> wrote:
> Anyone?

>  From FreeRadius log:

> Acct-Status-Type = Failed
> Service-Type = Sip-Session
> Sip-Response-Code = 408
> Sip-Method = Invite


> Event-Timestamp = "Jun 14 2011 16:02:47 CEST"
> Sip-From-Tag = "as16538a8e"
> Acct-Session-Id = "18375a0e058480f5626c94293975f7cf at x"


> User-Name = "a at x"
> Calling-Station-Id = "sip:a at x"
> Called-Station-Id = "sip:b at y"
> Sip-Translated-Request-URI = "sip:b at ip"


> Source-IP = "ip"
> Source-Port = "5060"
> Billing-Party = "sip:a at x"
> Canonical-URI = "sip:b at y"
> User-Agent = "Asterisk PBX"


> Contact = ""
> NAS-Port = 5060
> Acct-Delay-Time = 0
> NAS-IP-Address = 127.0.0.1
> Acct-Unique-Session-Id = "89f0c9014ba0a2f9"
> Timestamp = 1308060167


> Request-Authenticator = Verified

> Best regards,
> Tony Tyler


> 2011/6/14 Tony Tyler tonytyler2011 at gmail.com>


> Hi,

> We have setup CDRTool version 8.0.15 with an Asterisk multitenant PBX.



> We want to be able to log and bill the internal calls in CDRTool.
> All the calls are sent from Asterisk to OpenSIPS and the call is sent  
> back to the same Asterisk on the same IP-address.



> If it´sa 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.

>  From FreeRadius log:





> Acct-Status-Type = Failed
> Service-Type = Sip-Session
> Sip-Response-Code = 408
> Sip-Method = Invite
> Event-Timestamp = "Jun 14 2011 16:02:47 CEST"
> Sip-From-Tag = "as16538a8e"




> Acct-Session-Id = "18375a0e058480f5626c94293975f7cf at x"
> User-Name = "a at x"


> Calling-Station-Id = "sip:a at x"


> Called-Station-Id = "sip:b at y"
> Sip-Translated-Request-URI = "sip:b at ip"
> Source-IP = "ip"
> Source-Port = "5060"
> Billing-Party = "sip:a at x"




> Canonical-URI = "sip:b at y"
> User-Agent = "Asterisk PBX"
> Contact = ""
> NAS-Port = 5060
> Acct-Delay-Time = 0
> NAS-IP-Address = 127.0.0.1




> Acct-Unique-Session-Id = "89f0c9014ba0a2f9"
> Timestamp = 1308060167
> Request-Authenticator = Verified


> Best regards,
> Tony Tyler







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