Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

[Yves A. <yv...@gm...>]

OK,

if you can play DTMF on "normal" calls, but not on meetme conferences... 
have you tried to change the config parameter I recommended in my last 
eMail or
can´t you change this, because you want to avoid ANY changes in the 
config files on the target asterisk?

If you know that you are dialling into a meetme room (and I think you 
know, otherwise I would be totally confused..), you can dial directly 
into the
asterisk application meetme and pass the required parameters like 
roomnumber and pin... no need to play around with DTMF...

The following lines given as an example would establish a connection 
from the (SIP-)extension 567 to the meetme room 1024 that is protected with
the PIN 123456.

DefaultAsteriskServer asteriskServer = new 
DefaultAsteriskServer("###your asterisk server ip or dns-name###", 
"###username with appropriate rights###", "###password###);
asteriskServer.getManagerConnection().login();
asteriskServer.originateToApplication("SIP/567", "meetme", 
"1024,,123456", 15000l);

regards,
yves


Am 04.06.2013 14:39, schrieb Greg Horton:
> Hi Yves,
>
> The example I give is from a test system for my app, thus only 2 
> participant max.  I understand that is not a useful scenario.  In 
> short, our customer will have an existing Asterisk PBX which may have 
> been setup by FreePBX, TrixBox, manually, or anything else.  We are an 
> app that accesses that Asterisk system and cannot alter its 
> configuration.  In this case the customer will have configured 
> Asterisk to use Meetme and to require a PIN to enter the room.  Our 
> app can drive calls using Asterisk and A-J, but has no RTP/SIP (yet). 
>  The dialer in the GUI is a way to initiate calls from a known 
> Asterisk extension.  If I initiate a call to a known endpoint that has 
> an Asterisk channelID (other than Meetme) then I can use the dialer 
> send digits to Asterisk via PlayDtmfAction, and it works.  This could 
> be useful for something like prepaid cards, automated help desks, etc. 
>  When I call into the MeetMe conference room I just cannot get the 
> DTMF passed towards that channelID.  The Dialplan is waiting for a PIN 
> and I can't hook up with where it wants to read the PIN from (in 
> app_read.c).
>
> Maybe Asterisk's app_read can only get the PIN if it comes in an RTP 
> media stream (in-band or 2833) or via SIP Info method (or whatever IAX 
> does).  If that is the case there is nothing I can do at this time, 
> but I am just not sure it works that way.
>
> Thanks,
>
> Greg
>
>
> On Tue, Jun 4, 2013 at 2:44 AM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     hi,
>
>     as far as I remember, there is an option where you can set if dtmf
>     tones should pass the conference call... maybe this does the trick
>     for you.
>     if not..:
>     sorry, I still did not get it... dialer, gui, meetme..?? how does
>     this stick together? what is the sense of the conferenceroom with
>     a limit of two users?
>     if it is a dialerapplication, why not use direct bridging? (or, if
>     you have more than one agent and use some kind of predictive
>     dialing, you would normally
>     use queues...)
>     so... to be fully with you, I need to understand the whole
>     dialprocess as detailed as possible...
>     yves
>
>
>
>
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