[Freeswitch-users] freeswitch as a SIP Bridge
anthony.minessale at gmail.com
Fri Apr 4 18:55:37 PDT 2008
I believe so, visit us on irc and we'll figure it out.
On Fri, Apr 4, 2008 at 7:34 PM, Tim Meade <Tim.Meade at fusedware.com> wrote:
> Greetings all,
> I've just stumbled upon your project and it may solve an issue we are
> I've just spent about 3 weeks getting to know asterisk just to discover I
> don't think it can do what I need.
> We have a project where we have incoming calls on a SIP channel. We need
> to do a direct forward of these calls to an outgoing channel based to a
> number which is from our database. Simple to do in asterisk, but the
> problem is that we cannot have these calls "connected" between the two
> lines. They have an automated message at the beginning that is being
> activated when we do the answer before the dial of the second number in
> Out first idea is to bridge the incoming call directly to the outgoing
> call. The problem is that the incoming call cannot be "answered" and then
> we initiate the outgoing call. It needs to be a seamless bridge between the
> two calls. A nice feature would be to have a timer on the call. I saw a
> bounty for the timer feature, so I'm guessing (hoping) the bridging part can
> be done now.
> One other thought we are having is the ability to leave the incoming line
> "ringing" and dial the outgoing line until it is answered. At that time,
> answer the incoming and then bridge them together.
> So my question is: Can freeswitch do these things?
> Thanks and congratulations on the nice work!
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
Anthony Minessale II
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