[Freeswitch-users] freeswitch as a SIP Bridge

Tim Meade Tim.Meade at fusedWare.com
Sat Apr 5 08:58:14 PDT 2008

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.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Saturday, April 05, 2008 11:46 AM
To: freeswitch-users at 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 at fcnetwork.com<mailto:mcollins at fcnetwork.com>> wrote:


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! :)

Thanks for checking out FreeSWITCH!

-Michael, aka mercutioviz


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto: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<mailto: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!

Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org<mailto:Freeswitch-users at lists.freeswitch.org>

Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080405/e4c4dc8f/attachment-0002.html 

More information about the FreeSWITCH-users mailing list