[asterisk-users] Sip reinvite

[Vicky]

If canreinvite=yes is specified in sip.conf for 2 sip extensions and call
recording is disabled in asterisk, both legs have same codec  . Doesit
always does native bridging . I am
using freepbx . How can i know if a  call is going through asterisk or
they are bridged directly to each other ? Does sip reinvite gives
problems in billing ?
 Is there any cli command to know that ?
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