Asterisk provides support for SIP Session Timers (RFC 4028) through parameters in sip.conf. To set them in FreePBX you can use the advanced SIP settings. SST’s are supposed to provide a keep-alive mechanism, not a timer to end the call at a pre-defined duration! However, they quite often don’t work properly and cause calls to drop. The simplest fix is to disable them with “session-timers=refuse”. You may find my article here of interest:
http://kb.smartvox.co.uk/voip-sip/top-reasons-voip-calls-drop/