Nat options

I am running 11.02 asterisk and I am getting this error
sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead
Can somone point me witch file I can edit so I can change sip conf file permanently nat=force_rport,comedia. DO I need to edit source code ?

What version of FreePBX are you running?

i am using freepbx 2.11beta1.4

Use the SIP settings module. Check the current svn version of sip.conf for exact variable syntax.

Use the custom variable option in SIP settings module.

I changed module settings page.sipsettings.php and changed from yes to force_rport,comedia and I removed nat=no from extensions. Is there a way to prevent adding of nat=no to extensions. It seems that there must option not to include them at all. I now have external extensions having sound. I just have issue with incoming calls with no sound out going calls work fine

There are 2 files that should be modified:
for the extensions page in

and for asterisk sip Settings in

In both cases you should know what are you doing!