[asterisk-dev] Inquiry:Asterisk support KPML?

[Pavel Troller]

> On Wed, Jan 6, 2010 at 8:19 AM, Olle E. Johansson <oej at edvina.net> wrote:
> 
> >
> > 6 jan 2010 kl. 08.04 skrev hadi motamedi:
> >
> > > Dear All
> > > Can you please confirm if Asterisk supports KPML or not?
> > Not.
> >
> > Why would you need KPML? What's the application for it?
> >
> > For the rest: KPML is a standard for monitoring of DTMF keypresses in a
> > remote media stream.
> >
> > /O
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> 
> Sorry . I want to send the sip dialed digits in one-by-one digit basis . I
> mean if the subs dials 6659983 the dialed digits are being sent as
> 6,6,5,9,9,8,3 but not 6659983 in just one invite package . Can you please
> confirm if for such an application the KPML is really needed or not? How can
> I implement this scenario?

Hi!
  You don't need to use KPML for this, it works perfectly, if the SIP UA is
able to cooperate.
  This scenario can be implemented very easily:
  1) SIP UA calls a pre-defined SIP URI (typical example: sip:s at asterisk), most
     SIP UAs have a "hotline" feature allowing such a thing.
  2) An "s" extension is started on the Asterisk side, which can do any tricks
     required to receive further dialling (WaitExten(), Incomplete(), DISA(),
     Read()...). You can mix it with playing tones, announcements etc, or you
     can keep it silent to simulate a classic PBX behaviour.
  3) Once the dialling is complete, the extension logic routes the call out
     to some particular extension. 
  What you need, is to make asterisk to send 183 Session Progress with SDP,
thus establishing the media stream. A simple call to Progress() in the dialplan
makes it.
  Some UAs are not able to dial (via RFC2833 or INFO) prior answer (200 OK),
but most of them are. The only poor devices I know they cannot dial before
answer is a family of Gigaset VoIP/DECT phones, they have a stupid firmware,
they just emit a "can't do" sound when you try to dial onto a non-answered
call. 
  I hope this helps.
    With regards, Pavel.