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.
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