A2billing Sip Trunk Not Working

Hi Everyone,

I have configured asterisk server with a2billing in Debian server,when I try to add sip trunk from extension.conf that’s working fine.But In the case of a2billing its showing some issues,I have added the same sip trunk via a2billing and I make a call,at that time the sip trunk call showing like :

Using SIP RTP CoS mark 5
– Called

Problem is “|60|HRrL(10140000:61000:30000)91234567890” this entries appending to every call, so sip trunk providers rejecting the call, how can I remove this entries from the call?
Please help me

This is a forum on FreePBX, why would you ask the question here and not in the A2billing forum? I am very curious.

I have already posted the same in a2billing forum, but still not get a solution for this, So I posted here if any one know the solution that will more help for me.