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
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@gmail.com]
To: freeswitch-users@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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