[Freeswitch-users] sip_exclude_contact in mad boss scenario does not work

[Lesley Pervis]

I'm using a small variation on the mad boss extension in the default
dialplan, much like this, except instead of using groups, I'm using
multiple user/100X@${domain} lines.

http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix

The problem is that the sip_exclude contact does not work as I expect. I
expect that the conference is NOT bridged to the caller, but it is. I can
see in the logs that a conference leg to the calling phone is set up, and
the phone answers and puts it on hold. This means there are two conference
legs up on the calling phone: the one that initiated the page, and another
deaf leg. When the first leg hangs up, the deaf one is on hold, and to end
the conference, I have to resume that leg and hang it up.

Anyone know why this isn't working for me?

FreeSWITCH version: 1.0.head (git-d8d4d20 2012-03-14 19-00-26 +0000)

Thanks,
Les
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