Thanks Anthony.
I’m going to have to give it a shot. Kind of looking forward
to it.
The issue with asterisk, just to keep everyone up to speed, is with my DID
providers. We have three for testing: Broadvoice, VTWHITE, and DIDX.
When the call comes in, we do NOT want it answered. The
incoming call will always play an automated message and we do not want it to be
played until a live human being answers the forwarded call.
So the situation is:
Incoming call should be left to ring.
Outgoing call is initiated from our server to a known number.
When outgoing call is answered, we bridge the two: OR we just
bridge the incoming call to the outgoing while retaining control over the call.
A couple of people here have mentioned that it should be easy to
do.
Thanks.
Tim
From:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 05, 2008 11:46 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
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@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@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf
Of Tim Meade
Sent: Friday, April 04, 2008 5:34 PM
To: 'freeswitch-users@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
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