[Asterisk-Users] play message to callee beforeconnecttoincomingcall

C F shmaltz at gmail.com
Mon Jul 4 12:26:27 MST 2005


You start to not make any sense, you posted a question like this:

i try to do the following:

1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP Phone 100
5) the incoming call (at extension 999) should be connected to SIP Phone
100

any suggestions on how to implement this in an easy way?


Using queues it all 5 things will happen. So now you adding some new
stuff lets see.

On 7/3/05, Roland Zagler <r.zagler at fog.at> wrote:
> i had a look at the capabilities of queues and agents before
> and there are some missing points:
>
> -) the announcement has to be played to the caller until the callee
> answers (looped)

I don't see why you can't use queues because of this, this is exactly
what queues will do.

> -) the announcement has to start from the beginning (so i cannot use
> MoH)

Again queues will do this for you.

> -) due to the missing ability of needed priorisation of agents,

I thought you want to call one phone? Can you please explain what you
mean with 'priorisation of agents'? I think it's implemented in HEAD.

> i cannot use the agents feature as implemented in asterisk

Why not? for what you opened this thread it will do.

Do you have something against queues? it looks like you are trying to
avoid it. First you say realtime, which didn't tell me why you can't
use queues. Now you come up with something else. We are trying to help
you, but if you have a hard time taking help don't ask for help.

On 7/3/05, Roland Zagler <r.zagler at fog.at> wrote:
> i had a look at the capabilities of queues and agents before
> and there are some missing points:
> 
> -) the announcement has to be played to the caller until the callee
> answers (looped)
> -) the announcement has to start from the beginning (so i cannot use
> MoH)
> -) due to the missing ability of needed priorisation of agents,
> i cannot use the agents feature as implemented in asterisk
> 
> roland
> 
> -----Original Message-----
> From: C F [mailto:shmaltz at gmail.com]
> Sent: Monday, July 04, 2005 1:11 AM
> To: Roland Zagler
> Subject: Re: [Asterisk-Users] play message to callee
> beforeconnecttoincomingcall
> 
> I don't see why this doesn't work with realtime. The same it works
> with .conf files
> 
> On 7/3/05, Roland Zagler <r.zagler at fog.at> wrote:
> > Thanks for the suggestion, C F, but the problem is there is a rather
> big
> > database application behind with many users, so a static configuration
> > is not suitable for my needs. i am working mostly with realtime and
> agi.
> >
> > regards,
> > roland
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
> > Sent: Sunday, July 03, 2005 11:52 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] play message to callee
> > beforeconnecttoincomingcall
> >
> > I beleive queues will do it all for you.
> >
> http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf
> > Use this part to play the sound to the callers:
> > ........
> > ;queue-youarenext = "queue-youarenext" ; ("You are now first in
> line.")
> > ........
> > and use:
> > ........
> > ;announce = queue-markq
> > ........
> > To announce what you want to the callee
> >
> > Hope this helps
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> >
> >
> 
> 
>



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