[Freeswitch-users] freeswitch as a SIP Bridge

[Anthony Minessale]

having another look at your email now that I have more time to examine it. I
think that is relatively simple for us to do unless I am misunderstanding.

You want to make an inbound sip call that then results in an outbound call
and you do not want to connect the audio until the moment that the far end
party answers right?

With freeswitch you have many choices when originating and bridging a call
and it's possible to influence the behavior in many ways.


On Sat, Apr 5, 2008 at 2:03 AM, Michael Collins <mcollins at fcnetwork.com>
wrote:

>  Tim,
>
>
>
> I can attest to the fact that FreeSWITCH is a great project with a really
> cool community.  Right now the IRC channel is the best place to get quick,
> specific information on how to handle the kinds of things you're looking to
> do.  One of the important things about FS is that the devs made extremely
> wise engineering decisions long before a single line of code was written.
> They made sure that FS would be extremely flexible in what it can do.  One
> byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do
> this?" when no one here had thought of that before.  In many cases it can
> indeed "do that," but it takes a little tweaking and some attention from the
> experts.  Definitely hop on the IRC channel #freeswitch; the main devs
> generally are there during the day and some evenings.
>
>
>
> If you do get FS up and running, especially in a production environment,
> we would ask that you consider documenting your setup on the wiki.  We hope
> to lower the barrier to entry for others with similar needs by having good
> documentation, but of course we also like to brag about what FS can do! J
>
>
>
> Thanks for checking out FreeSWITCH!
>
>
>
> -Michael, aka mercutioviz
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
> *Sent:* Friday, April 04, 2008 5:34 PM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* [Freeswitch-users] freeswitch as a SIP Bridge
>
>
>
> 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

FreeSWITCH http://www.freeswitch.org/
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