[asterisk-dev] Best practice requested
Greg Boehnlein
damin at nacs.net
Mon Oct 2 17:05:42 MST 2006
yOn Mon, 2 Oct 2006, Jeremy McNamara wrote:
> Mitch Sharp wrote:
> > My question is, once the conference room has been created, what is the
> > best practice method to unbridge SIP/101 and Zap/1 in order to then move
> > them into the conference room.
>
> Why not use a conference for every call? This way the manager can come
> in and out whenever they needed. If you set the dialplan up properly
> the manager could come in muted, then un-mute at any time to take over
> the call.
>
> We have used this approach in many situations with great success.
Would that have any potential DTMF pass-through issues? It sounds workable
for something a customer of mine is asking for, but they need to have full
DTMF access to both sides of the channel.
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
More information about the asterisk-dev
mailing list