[Freeswitch-users] freeswitch as a SIP Bridge

[Anthony Minessale]

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
> having.
>
>
> 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
> asterisk.
>
>
>
> 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!
>
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>
>


-- 
Anthony Minessale II

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