But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed.
Capture a trace of the call using Wireshark. Are you sure there's no RTP packets during that period? Also remember that the config change needs to be picked up by restarting the FreeSWITCH profile (the easiest way to be 100% sure is just to restart FS). Sharing your configuration could also be useful, in case there's a typo.
Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities!
Correct, they're different. Session timers are a request/response in the SIP signalling layer. RTP timers are simply no RTP media packets received for a given period of time. There's no dependancy between them. You can use either none/one/the other/both. Incidentally the reason to prefer SIP session timers to RTP timers is that not all calls will send RTP. For example FS won't by default when it's only recording something (voicemail): http://wiki.freeswitch.org/wiki/Variable_record_waste_resources -Steve