[Freeswitch-users] Non Blocking Teletone (Background Audio)

Anthony Minessale anthony.minessale at gmail.com
Thu Jun 26 08:02:00 PDT 2008


try to replace "ring_ready" with "pre_answer"

if your phone does not support early media you will not be able to generate
inband indication.

ring_ready is a an app that sends a out of band ringing indication so in the
case of SIP it
will generate a 180 ringing which will make most devices send a ringback
sound.


On Mon, Jun 23, 2008 at 8:38 PM, Klaus Teller <klaus.teller at gmx.net> wrote:

> I'm trying the third variant without much success. Here is what i do:
>
>
> session.execute("ring_ready");
> session.execute("set","ringback=%(2000,4000,440,480)");
> session.execute("set", "transfer_ringback=%(2000,4000,440,480)");
>
> The main problem i have is that there is no tone at all. No dial tone, and
> no ring tone. Any wrong with this code?
>
> Thanks,
> Klaus L'Imbecile.
>
>
> -------- Original-Nachricht --------
> > Datum: Mon, 23 Jun 2008 08:45:56 -0500
> > Von: "Anthony Minessale" <anthony.minessale at gmail.com>
> > An: freeswitch-users at lists.freeswitch.org
> > Betreff: Re: [Freeswitch-users] Non Blocking Teletone (Background Audio)
>
> > In order to do 2 things at once like that you need 2 threads.
> > Here are your options:
> >
> > 1) Use outbound mode of mod_event_socket to connect to a tcp port and
> > control the call from there.
> > 2) Use the jsrun api call to launch another js in it's own thread and
> then
> > uuid_bridge the channels.
> > 3) *Easiest* set the ringback variable on the A leg to the teletone
> script
> > of choice or path to an audio file so when you dial B the A will hear it.
> >
> > On Sun, Jun 22, 2008 at 9:49 PM, Klaus Teller <klaus.teller at gmx.net>
> > wrote:
> >
> > > Hi Folks,
> > >
> > > I was was wondering if it's possible to play a teletone in a
> > non-blocking
> > > way.
> > >
> > > What I want to achieve is following:
> > >
> > > 1) Party A calls to my FS box,
> > > 2) A Javascript script picks up the call
> > > 3) The script generates a teletone to party A.
> > > 4) Then the script starts some processing while A is on hold listening
> > for
> > > the teletone (or any other audio?)
> > > 5) The script dials some other party B
> > > 6) Party B picks up
> > > 7) Script stops the teletone being played to A
> > > 8) Script bridges A and B.
> > >
> > > As far as I could experiment with teletone.generate, the execution of
> > the
> > > script is paused when the teletone is played to the near end (Party A).
> > >
> > > I would love if you could help me with this.
> > >
> > > Thanks,
> > > Klaus L'Imbecile.
> > >
> > >
> > > --
> > > Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!
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> > >
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> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
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> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
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> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
>
> --
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> Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer
>
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
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