Username

Main user of the SIP user that started the call.

Sip user

SIP user that requested the call.

Credit

User credit.

Number

Dialed number.

Codec

Codec used.

CallerID

The CallerID number.

Trunk

Trunk that was used to complete the call.

Reinvite

Reinvite indicates whether the audio is passing through Asterisk or directly between the client and the trunk. You can see more details at the link https://wiki.magnusbilling.org/en/source/asterisk_options/directmedia.html..

From IP

IP of the caller.

Description

Data from the sip show channel command.