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

[Michael Collins]

Grab a console debug log and throw it on pastebin.freeswitch.org. Use
"FreeSWITCH Log" for syntax highlighting and then give us the pb URL in
this thread.
-MC

On Thu, Mar 29, 2012 at 5:02 PM, Lesley Pervis <lesley.pervis at gmail.com>wrote:

> 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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