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 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061125/99701438/attachment.htm