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<div class="gmail_quote">On Wed, Jan 6, 2010 at 10:45 AM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net">oej@edvina.net</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"><br>6 jan 2010 kl. 11.24 skrev hadi motamedi:<br>
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<div class="h5"><br>><br>><br>> 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>> 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 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>> 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?<br>
<br></div></div>That's not KPML, that's overlap dialing where you send 6 in the first invite, get a negative reply, then send 66, then 665 etc until you get a positive reply.<br>
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<div class="h5"><br>/O<br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
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<div>Thank you very much for your reply . Can you please let me know how can I find an example of this implementation where my Asterisk has sip connection to an external sip server and I need to send my subs dialed digits toward the external sip server in one-by-one digit basis ?</div>
<div>Thank you in advance</div>
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