You can enable SRTP from your Dialplan on a per-call basis by setting the following flag:

<action application="set" data="sip_secure_media=true"/> 

This needs to be done on both legs and on both inbound and outbound calls to be fully effective. Of course, the ITSP that provide you gateway to PSTN will probably not support SRTP so you may only be able to enable this on legs from FreeSWITCH to the endpoint(s).

You can check if media is secured properly by checking for the variable ${sip_secure_media_confirmed} to be set. As an example, the following block will play a bong tone when SIP media is secured:

<extension name="is_secure"> 
<condition field="${sip_secure_media_confirmed}" 
expression="^true$"> 
<action application="sleep" data="1000"/> 
<action application="gentones" data="${bong-ring}"/> 
</condition> 
</extension>