[asterisk-users] playing a sound into a meetme conf

Simon Austin simon at openflows.org
Fri Jul 28 12:20:43 MST 2006


Thank you.

I had previously seen the Local channel channel but I didn't completely
understand how it worked.  That was a really good explanation of how I can
do it.

Cheers,

- Simon

On 7/27/06, Moises Silva <moises.silva at gmail.com> wrote:
>
> may be im missing something, but i think the pseudo channel you are
> looking for is called "Local" and you can call some extension that you
> know the only thing it does is play the message you need. So you can
> originate a call to that Local channel and bridge it to the Meetme
> conference where your users are waiting.
>
> [meetme-play-message]
> exten => s,1,Answer()
> exten => s,2,Playback(were_sorry_the_translator_is_gone)
> exten => s,3,Hangup()
>
> [receive-meetme-message]
> exten => s,1,Answer()
> exten => s,2,AGI(some.pl) /* assuming is needed to execute agi to know
> wich conference to join*/
> exten => s,3,Meetme(${variable_conference_set_from_perl_agi})
> exten => s,4,Hangup()
>
> then, when a translator is gone, from the DeadAgi execute a manager
> action "Originate" to call Local channel
>
> Action: Originate
> Channel: Local/s at meetme-play-message
> Context: receive-meetme-message
> extension: s
> priority: 1
>
>
> You can do it in both directions :)
>
> It would be enough?
>
> Regards
>
> On 7/27/06, Simon Austin < simon at openflows.org> wrote:
> > Hi All,
> >
> > I have a problem and I'm not sure if a solution is possible without
> using
> > the asterisk testing code.
> >
> > I am developing a volunteer translation service that users can dial
> into.
> > I have a list of volunteer translators cell phone numbers stored in a
> > mysql database along with times that they have volunteered to act as
> > translators.  That I pull from using some perl AGI scripts.
> >
> > A user calls, I ask which language they need help/translation with, then
> I
> > put the users into a meetme conference while I call translators and play
> > them a message asking if they're available at this time.  They can
> refuse
> > or accept the call.
> >
> > Once I get a translator that has accepted the call I connect the
> > translator as an administrator to the meetme conference that is holding
> > the user that is listening to music on hold.
> >
> > That is all working quite well with the Dialplan and AGI scripts I have
> > set up.
> >
> > Problems happen when the translator drops the call midway through the
> > conversation.  i.e. Losing cell phone service.
> >
> > When that happens I need a way to play a message to the user to let them
> > know that the translator has been lost and we're looking for a new one.
> >
> > I then need to put back the music on hold, then run deadagi scripts to
> > find a new translator to connect to the meetme conference to help out
> the
> > user.
> >
> > What is currently happening is that the user is left in the conference
> > alone forever listening to MOH.
> >
> > I think there are two ways to do this, but I can't find out how to do
> > either from any documentation I've found.
> >
> > 1. Break the user out of the meetme conf and back into the dialplan.
> >   - If I kick them from the conference they are immediately hung up on
> and
> > I don't know how to stop this from happening.
> >   - There is function that is available in Asterisk 1.4 called
> > ManagerRedirect that seems like it could do this for me, but i'd rather
> > not try to integrate this into 1.2.10 because I fear breaking too many
> > other things and running 1.4 (testing) just isn't an option at this
> time.
> > (details here: http://bugs.digium.com/view.php?id=6508)
> >
> > 2. Play a message into the conference
> >   - Can I join a new pseudo channel that I've created to a meetme conf
> > that plays a message?  Does anyone know how to do this?
> >   - Can I override the MOH and stream a recorded message into the
> > conference with only the single user in the meetme conf?
> >
> > Any help/ideas are appreciated.
> >
> > Cheers,
> >
> > - Simon
> >
> > Simon Austin
> > http://simon.openflows.org
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