[Freeswitch-users] Conference dialing and uuid

Tue Feb 3 00:25:30 PST 2009

Hmmm no MOH wont work... since I am planning on pulling more than just 2
members into the conference and I still need ringback for the later
members as well.

Is there a direct way for me to use conference <number> play <audiofile>
to play teletone directly? or should I just records some ringing if I
want to use that?

And lastly for my own sanity ;-) why would the following in a
on_ring_execute stop execution of the call at that point?

call = argv[1];
conf = argv[2];

consoleLog("info","Making ringback channel for uuid : "+ session.uuid
+"\n");
var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf +"-conf &park()")

//I tried with and without a exit() at the end

It seems to stop media detection??(not really sure about the term) for the call that executes this
script.

Freeswitch doesent recognize the pickup of that call and thus it doesent
get bridged into the conference. when I uuid_kill the call that gets
originated everything else starts happening again. 

Oh Im running this in FS ver. 1.0.trunk (11226:11561M)

and that loopback points to 
<extension name="ringback">
<condition field="destination_number" expression="^ringback-conf=(.*)$">
<action application="javascript" data="ringback.js $1"/>
</condition>
</extension>

and ringback.js is

use("TeleTone");  
session.answer();  
var tts = new TeleTone(session);

tts.addTone("u", 400.0, 450.0, 0.0);
tts.addTone("r", 440.0, 480.0, 0.0);

var RESET = "v=2000;>=0;+=0;";
var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
var US_RING = RESET + "r(2000,4000)";

while(session.ready()) {    
  console_log("making UK ring\n");
    for (x = 0 ; x < 2 ; x++) {
        tts.generate(UK_RING);
    } 
} 

A slight bastardisation of the teletone JS example.

I would expected the new channel that is created via a api originate to
be completely seperate from the JS I create it in. (thats why I use api
instead of creating a new session, although I should probably try that
as well).

I use some CoreDB stuff to keep tabs on the uuid for the originated call
so that I can uuid_kill it in the on_answer_script but as mentioned...
the on_answer only executes after I uuid_kill the originated channel in
the cli... 

Thanks again guys,
Specially since it seems you two are always the ones that get back to
me.

On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote:
> Actually loopback does work.
> however as I said it generates a pair of extra channels.
> 
> Hmmm I was trying to generate and extra call to a JS script that
> generated a teletone ring in an on_ring_execute for the second call
> however it seems to stop execution of the call itself. Event though I
> use api commands to originate and then transfer it into the conference
> so that I have direct access to its uuid.
> 
> I think changeing the moh might be a bit simpler however and elimite
> some CoreDB stuff I was doing to keep track of the calls ring generating
> call (what a sentance).
> 
> On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote:
> >    you could set the conference moh sound to be tone_stream::// with the
> >    teletone spec for ring sound and it use ignore_early_media=true in your
> >    originates so the first caller would hear ringback until the 2nd one
> >    arrived.
> > 
> >    On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org>
> >    wrote:
> > 
> >      Loopback will not work in that case either.  If the far end plays
> >      ringback inband you should hear that if you use the conference dial
> >      api call.
> >      /b
> > 
> >    On Feb 2, 2009, at 4:24 AM, Sias Mey wrote:
> >    > Aaah ok.
> >    >
> >    > Thanks for clearing that up.
> >    >
> >    > So using loopback is still the only real workable sollution for me,
> >    > since that generates ringback from and alternative endpoint and
> >    > plays it
> >    > into the conference.
> >    >
> >    > I might play with some javascript that streams ring into the channel
> >    > eventually but for now the string comparisons at least get me the
> >    > right
> >    > uuid.
> >    >
> >    > Thank you again,
> >    > Sias
> > 
> >    _______________________________________________
> >    Freeswitch-users mailing list
> >    [2]Freeswitch-users at lists.freeswitch.org
> >    [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >    UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u
> >    sers
> >    [5]http://www.freeswitch.org
> > 
> >    --
> >    Anthony Minessale II
> >    FreeSWITCH [6]http://www.freeswitch.org/
> >    ClueCon [7]http://www.cluecon.com/
> >    AIM: anthm
> >    [8]MSN:anthony_minessale at hotmail.com
> >    GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com
> >    IRC: [10]irc.freenode.net #freeswitch
> >    FreeSWITCH Developer Conference
> >    [11]sip:888 at conference.freeswitch.org
> >    [12]iax:guest at conference.freeswitch.org/888
> >    [13]googletalk:conf+888 at conference.freeswitch.org
> >    pstn:213-799-1400
> > 
> > References
> > 
> >    1. mailto:brian at freeswitch.org
> >    2. mailto:Freeswitch-users at lists.freeswitch.org
> >    3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >    4. http://lists.freeswitch.org/mailman/options/freeswitch-users
> >    5. http://www.freeswitch.org/
> >    6. http://www.freeswitch.org/
> >    7. http://www.cluecon.com/
> >    8. mailto:MSN%3Aanthony_minessale at hotmail.com
> >    9. mailto:PAYPAL%3Aanthony.minessale at gmail.com
> >   10. http://irc.freenode.net/
> >   11. mailto:sip%3A888 at conference.freeswitch.org
> >   12. http://iax:guest@conference.freeswitch.org/888
> >   13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
> 
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> 
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