[Asterisk-Users] Call screening

[Steven Critchfield]

In the case of Sip, or H323, you can not be certain the audio is going
through asterisk. If it is Sip-Sip, or h323-h323, the audio should go
directly from one side to the other and not traverse asterisk. In this
scenario, a barge function could not get into the stream since it is not
part of the pathway.

On Mon, 2003-04-21 at 13:04, Steve Radich wrote:
> Has anyone else wondered why we can't do a SIPBarge, H323Barge, etc. 
> 
> I took an extremely brief look at the code and it looks like it's probably
> zap specific - but is a barge app for the other interfaces something anyone
> else is writing?
> 
> Steve Radich
> BitShop, Inc.
> 
> -----Original Message-----
> From: Brancaleoni Matteo [mailto:mbrancaleoni at espia.it]
> Sent: Sunday, April 20, 2003 5:02 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Call screening
> 
> 
> If you're using zap channels
> you could use the Zapbarge app.
> Just set up an extension to start
> the app, like
> 
> exten => 8000,1,ZapBarge
> exten => 8000,2,Hangup
> 
> Asterisk then will prompt for a chan
> number (zaptel only!) and let
> you listen to what passes
> on that channel...
> 
> You can barge from any channel, but
> only into zaptel (ie you can listen
> from sip to a zap channel).
> 
> Matteo
> 
> Il dom, 2003-04-20 alle 01:07, Joel Scotkin ha scritto:
> > I've set up asterisk with my X100P as a home answering machine.  Works
> great
> > so far - answers the phone after 20 seconds, runs the phone tree, emails
> > voicemail, etc.
> > 
> > However, the one feature traditional answering machines have that I
> haven't
> > been able to figure out is how to listen in on the call.  Ideally I could
> > just route through Console/dsp and hear it on my speakers.  I've tried the
> > monitoring app, but that only seems to log to files.  I couldn't figure
> out
> > any way to set up a meetme conference while still running through the
> > various stages in answering the call.  Any ideas?
> > 
> > Things I've considered, but maybe didn't drill down far enough:
> > 
> > - Unload chan_oss, and get the monitoring app to just write to /dev/dsp.
> Or
> > maybe to a pipe, which I can then pass through the standard linux "play"
> > application.  res_listen anyone?
> > 
> > - Figure out meetme for this purpose.
> > 
> > - Three way call to the console?  I couldn't figure this one out either.
> > 
> > Any of these solutions bring up the related question - is there any way to
> > get Record to hang up automatically if any other (non-asterisk connected)
> > extension is picked up in the house?  Traditional answering machines
> detect
> > this, but I'm not sure how.  The easy workaround is simply to hit the #
> key
> > on the local extension to force a hangup.
> > 
> > I'd appreciate any insights (or code!).  Thanks!
> > 
> > Joel
> > 
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
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> 
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-- 
Steven Critchfield  <critch at basesys.com>