Hi I have a FreeSWITCH install that uses a Faktortel SIP gateway for incoming/outgoing calls. The gateway codec is g.722 and everything works fine except for redirection to voicemail. (I have hard coded the codec for the gateway since it would not work otherwise.) Here is the conf/dialplan/public/01_line1.xml file with the dialplan for incoming calls <extension name="line1"> <condition field="destination_number" expression="^(5555555555)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="group/sales@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain} 1000"/> </condition> </extension> When the call is redirected to voicemail, half of the playback message is lost and the first thing the user hears is "to listen to the message press 1..." If I change the dialplan to transfer to the "1000" extension instead, allowing FreeSWITCH to redirect to voicemail automatically when ext 1001 does not answer, then voicemail works correctly. Any suggestions? Thanks O.