[asterisk-dev] Inquiry:Asterisk support KPML?

[hadi motamedi]

On Wed, Jan 6, 2010 at 11:02 AM, Pavel Troller <patrol at sinus.cz> wrote:

>  > 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.
>
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Thank you very much for you help . My Asterisk has sip connection with an
external sip server that I need to send my subscriber dialed digit toward
the external sip server in one-by-one digit basis . Can you please confirm
if the following code can fulfill the job ?
Under extensions.conf :
---------------------------------
exten => _X!,1,macro(dialsip,SIP/osaka/${EXTEN})
[macro-dialsip]
exten => s,1,Dial(${ARG1})
exten => s,n,WaitExten()
 exten => s,n,Dial(${ARG1})
exten => s,n,WaitExten()
 exten => s,n,Dial(${ARG1})
exten => s,n,WaitExten()
 exten => s,n,Dial(${ARG1})
exten => s,n,WaitExten()
 exten => s,n,Dial(${ARG1})

Thank you
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