[asterisk-dev] Bridging two Channels

[ast guy]

Discussion is about application development,  IMHO developers are more aware
of * API than normal * users. Thanks for sighting app_bridge, I have read
about it and comes with *-1.6-beta, but I have option 1 as 1.2 and option 2
for 1.4. So can I do such trick in *-1.2. I think I need to go through it's
code implementation.

-ag

On Sat, Apr 26, 2008 at 5:41 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> 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
> >
>
> _______________________________________________
> --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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080426/5e5a2a87/attachment.htm