[asterisk-dev] Bridging two Channels

[Steve Totaro]

This is really not a Dev question but a users question.  At the risk
of encouraging posting to the incorrect list I will give you a hint.
Google app_bridge.

Thanks,
Steve Totaro

On Sat, Apr 26, 2008 at 7:18 AM, ast guy <astguy at gmail.com> wrote:
> Well I'm expecting around 30-40 concurrent calls, 80 channels in total.
>
> -ag
>
>
>
> On Sat, Apr 26, 2008 at 2:14 PM, Wolfgang Pichler <wpichler at yosd.at> wrote:
>
> > Hi,
> >
> > i think the best way (maybe the only way - i don't know exactly) would
> > be to use the manager command redirect and redirect both channels into a
> > conference (i don't think that you have that much overhead there - how
> > many channels at the same time will do that ?)
> >
> > regards,
> > Wolfgang
> >
> > ast guy schrieb:
> >
> >
> >
> > > Hi,
> > >  I'm looking for some approach where I can bridge two different
> > > channels. Let me explain the scenario.
> > > channel-A lands in dial plan and executes an application-X. Now there
> > > is another channel-B in the same context but on different application
> > > say Playback() . What is the best approach to bridge both channels?
> > >
> > >  - Add both channels in conference ? Is a good approach, what about
> > > resource usage ?
> > >  - Any code/API available to do bridge both, like native pbx behavior ?
> > >
> > > If both channels have been bridged then will channel-A return to
> > > application-X ? and channel-B to Playback() ? after bridge is no
> longer...
> > > Well I'm also interested in to hangup channel after a specific time
> > > out value has reached or either party hangs up.
> > >
> > >
> > > -AG
> > > ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> > >
> > > asterisk-dev mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
>
> _______________________________________________
>  --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
>  asterisk-dev mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>