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<p class=MsoNormal>Greetings all,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’ve just stumbled upon your project and it may solve
an issue we are having. <o:p></o:p></p>
<p class=MsoNormal><br>
I’ve just spent about 3 weeks getting to know asterisk just to discover I
don’t think it can do what I need.<o:p></o:p></p>
<p class=MsoNormal><br>
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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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. <o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>So my question is: Can freeswitch do these things?<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Thanks and congratulations on the nice work!<o:p></o:p></p>
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