[Freeswitch-users] freeswitch as a SIP Bridge

[Tim Meade]

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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