[Freeswitch-users] Reject incoming call (namely session) on conference member limit

[Daniel Eiland]

Awesome thanks, just the direction I needed.  Still trying to understand
the dial plan and everything else.

For anyone else wondering, I created the following dial plan conditions and
it seems to do the trick:

<extension name="meetme">
    <condition field="destination_number" expression="(^MEETME\d\d)"
break="on-false">
        <action inline="true" application="set"
data="member_count=${conference( $1 get count)}"/>
        <action inline="true" application="set"
data="member_limit=${sofia_gateway_data( $1 var member_limit)}"/>
        <action inline="true" application="set" data="dialed_ext=$1"/>
    </condition>
    <condition field="${cond(${member_count} == ${member_limit} ? YES :
NO)}" expression="^YES$" break="on-true">
        <action application="respond" data="486"/>
    </condition>
    <condition>
        <action application="answer" data="is_conference"/>
        <action application="conference" data="${dialed_ext}@default"/>
    </condition>
</extension>

-Daniel



On Tue, Oct 9, 2012 at 3:24 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> Check out the wiki to see the conference API, check the count of members
> on a conference using that API from the dialplan, and if it's at the max
> then use Application=respond, data=486, to blow off the call with a busy.
>
> Sorry I don't have the details for you, but I wanted to point you in the
> right direction ASAP.
>
> --Dave
>
>  ------------------------------
> *From:* Daniel Eiland [mailto:daniel.eiland at gmail.com]
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tue, 09 Oct 2012 11:27:12 -0700
> *Subject:* [Freeswitch-users] Reject incoming call (namely session) on
> conference member limit
>
>
> Hey folks,
>
> I was poking around the mod_conference settings and saw that I can limit
> the maximum members in a conference (via the max-members parameter).
> However, once this limit is reached, the next incoming call is still
> answered and played a tone (designated via max-members-sound) before it is
> simply hung-up.
>
> In the SIP world, this looks like an INVITE-200-ACK sequence followed by a
> BYE-200 sequence.  Is there anyway I can skip the BYE-200 sequence and
> simply reject the call outright with say a 486 response to the INVITE?
>
> Thanks,
> Daniel
>
>
>
>
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