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<div class="gmail_quote">On Wed, Jan 6, 2010 at 11:02 AM, Pavel Troller <span dir="ltr"><<a href="mailto:patrol@sinus.cz">patrol@sinus.cz</a>></span> wrote:<br>
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<div class="h5">> On Wed, Jan 6, 2010 at 8:19 AM, Olle E. Johansson <<a href="mailto:oej@edvina.net">oej@edvina.net</a>> wrote:<br>><br>> ><br>> > 6 jan 2010 kl. 08.04 skrev hadi motamedi:<br>> ><br>
> > > Dear All<br>> > > Can you please confirm if Asterisk supports KPML or not?<br>> > Not.<br>> ><br>> > Why would you need KPML? What's the application for it?<br>> ><br>> > For the rest: KPML is a standard for monitoring of DTMF keypresses in a<br>
> > remote media stream.<br>> ><br>> > /O<br>> > _______________________________________________<br>> > --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
> ><br>> > asterisk-dev mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>
> ><br>><br>><br>> Sorry . I want to send the sip dialed digits in one-by-one digit basis . I<br>> mean if the subs dials 6659983 the dialed digits are being sent as<br>> 6,6,5,9,9,8,3 but not 6659983 in just one invite package . Can you please<br>
> confirm if for such an application the KPML is really needed or not? How can<br>> I implement this scenario?<br><br></div></div>Hi!<br> You don't need to use KPML for this, it works perfectly, if the SIP UA is<br>
able to cooperate.<br> This scenario can be implemented very easily:<br> 1) SIP UA calls a pre-defined SIP URI (typical example: sip:s@asterisk), most<br> SIP UAs have a "hotline" feature allowing such a thing.<br>
2) An "s" extension is started on the Asterisk side, which can do any tricks<br> required to receive further dialling (WaitExten(), Incomplete(), DISA(),<br> Read()...). You can mix it with playing tones, announcements etc, or you<br>
can keep it silent to simulate a classic PBX behaviour.<br> 3) Once the dialling is complete, the extension logic routes the call out<br> to some particular extension.<br> What you need, is to make asterisk to send 183 Session Progress with SDP,<br>
thus establishing the media stream. A simple call to Progress() in the dialplan<br>makes it.<br> Some UAs are not able to dial (via RFC2833 or INFO) prior answer (200 OK),<br>but most of them are. The only poor devices I know they cannot dial before<br>
answer is a family of Gigaset VoIP/DECT phones, they have a stupid firmware,<br>they just emit a "can't do" sound when you try to dial onto a non-answered<br>call.<br> I hope this helps.<br> With regards, Pavel.<br>
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<div class="h5"><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>
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<div>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 ?</div>
<div>Under extensions.conf :</div>
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<div>exten => _X!,1,macro(dialsip,SIP/osaka/${EXTEN})</div>
<div>[macro-dialsip]</div>
<div>exten => s,1,Dial(${ARG1})</div>
<div>exten => s,n,WaitExten()</div>
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<div>exten => s,n,Dial(${ARG1})</div>
<div>exten => s,n,WaitExten()</div>
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<div>exten => s,n,Dial(${ARG1})</div>
<div>exten => s,n,WaitExten()</div>
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<div>exten => s,n,Dial(${ARG1})</div>
<div>exten => s,n,WaitExten()</div>
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<div>exten => s,n,Dial(${ARG1})</div>
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<div>Thank you</div>
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