[Freeswitch-users] caller control options for outbound conference calls

[Ketan Kothari]

Hello there,

We are trying to use caller control options for outbound conference calls.
We are using originate command to originate conference call with PSTN
numbers.
I am using mod_XML for write conference.
I use default caller controls as well as also added custom caller controls.
My caller control name is plain. I have added it in dialplan parameter
conference_control=plain as well as add it in conference parameter still
caller mute option is not working.
Below is my conference XML:

<document type="freeswitch/xml">
   <section name="Configuration" description="Configuration">
       <configuration name="conference.conf" description="Audio Conference">
        <advertise>
                <room name="0676338702 at xxx.xxx.xxx.xx" status="FreeSWITCH"/>
        </advertise>
        <caller-controls>
        <group name="plain">
                <control action="mute"  digits="0"/>
                <control action="vol talk dn"  digits="1"/>
                <control action="vol talk zero"  digits="2"/>
                <control action="vol talk up"  digits="3"/>
                <control action="vol listen dn"  digits="4"/>
                <control action="vol listen zero"  digits="5"/>
                <control action="vol listen up"  digits="6"/>
                <control action="energy dn"  digits="7"/>
                <control action="energy equ"  digits="8"/>
                <control action="energy up"  digits="9"/>
                <control action="deaf mute"  digits="*"/>
                <control action="execute_application"  digits="#"
data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default" />
        </group>
        </caller-controls>
        <profiles>
                <profile name="test">
                        <param name="caller-id-number" value="xxxxxxxxxx"/>
                       <param name="enter-sound"
value="tone_stream://%(200,0,500,600,700)"/>
                       <param name="exit-sound"
value="tone_stream://%(500,0,300,200,100,50,25)"/>
                        <param name="auto-record"
value="${recordings_dir}/testrushika_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
                        <param name="member-flags" value="dist-dtmf"/>
                        <param name="pin-retries" value="10"/>
                        <param name="max-members" value="1000"/>
                        <param name="status" value="0"/>
                       <param name="caller-controls" value="plain"/>
                </profile>
       </profiles>
       </configuration>
   </section>
</document>

Please let me know if anything is not correct or any new configuration need
to go.
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