Well... I got it!!!! Not sure if there are any bugs yet or for a safe
stand point. Or if this is everything I tried a lot of stuff. But for
the newbie like me. If you want to use free switch to dial local sip
extensions here is how. Add the Hash Mod to your auto_load Configs.
and to your modxml file. Then add the following lines of code to your
default dial plan
<!--
dial the extension (1000-1019) for 30 seconds and go to voicemail if
the
call fails (continue_on_fail=true), otherwise hang up after a
successful
bridge (hangup_after_bridge=true)
-->
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])
$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app>
-->
<action application="bind_meta_app" data="1 b s execute_extension::dx
XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$$
{recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-
%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf
XML features"/>
<action application="bind_meta_app" data="4 b s
execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=$
{network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT ,NO_ROUTE_DESTINATION"/
> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/$
{dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/$
{dialed_extension}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/$
{called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/
global/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data($
{dialed_extension}@${domain_name} var callgroup)}"/>
<!--<action application="export" data="nolocal:sip_secure_media=$
{user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/
>-->
<action application="hash" data="insert/${domain_name}-last_dial/$
{called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@$
{domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default $
{domain_name} ${dialed_extension}"/>
</condition>
</extension>
<!-- END freeswitch Stuff -->
<!-- BBB Stuff -->
<extension name="bbb_conferences">
<condition field="destination_number" expression="^(\d{5})$">
<action application="answer"/>
<action application="conference" data="$1@wideband"/>
<!-- <action application="conference" data="$1@wideband"/> --
</condition>
</extension>
<!-- END BBB Stuff -->
Now you need your sip phones to connect to your freeswitch internally.
So have your phones connect to port 5090. So use <serverip>:5090. That
should be it. refresh your xmls and clean bbb and should start up.
Then dial the default extensions or your 5 number bbb ID
