[freeside-users] Freeside+Voip (Sip protocol)

[harry gaillac]

Hello,

I wish to install freeside to test voip billing (SIP).
I use SER and asterisk.

SER is able to support accounting (no b2bua) .


Ua-----|SER|-------Asterisk---PSTN
         |
     freeradius/mysql

When SER receive an INVITE message from Ua this one is
sent to asterisk and radacct is filled out with unique
id and start time for the session.

Ua send back ACK message to Asterisk for the session
so SER fill out the radacct table with  the same
unique id and the start time.

When Ua send Bye message SER fill out the radacct
table with  the same unique id and the stop time.

How freeradius calculate the time session between ACK
and BYE message.VoIP billing is calculated in
FS/FS/part_pkg/voip_sqlradacct.pm and the 
radacct table is queried in
FS/FS/part_export/sqlradius.pm.

Does these script provide the solution?


VoIP billing is calculated in
FS/FS/part_pkg/voip_sqlradacct.pm and the 
radacct table is queried in
FS/FS/part_export/sqlradius.pm.

Regards

Harry



	

	
		
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