<div>Hello!</div><div><br></div>For ASR menus check this out: <a href="http://wiki.freeswitch.org/wiki/Mod_pocketsphinx">http://wiki.freeswitch.org/wiki/Mod_pocketsphinx</a><div><a href="http://wiki.freeswitch.org/wiki/Mod_pocketsphinx"></a>I wouldn't say it is perfect but it is nice, on the other hand there is this dialplan tool that makes fs detect inband dtmf: <a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf</a></div>
<div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf"></a>hope it helps.</div><div><br></div><div>-Milena<br><br><div class="gmail_quote">2010/2/17 Doc <span dir="ltr"><<a href="mailto:ledoktre@meanie.us">ledoktre@meanie.us</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Once you mentioned the in-band versus out-of-band DTMF, it made sense.<br>
I've been reading on it since your post, and I'm going to try a few<br>
things. Wonder if there is any app for voice controlled IVR (so many<br>
systems seem to support it these days, it would be a nice way to<br>
circumvent the DTMF issue... :-) ).<br>
<br>
Any ideas on the rest of my points? -Thanks,<br>
<div><div></div><div class="h5"><br>
Scott Torr wrote:<br>
> On Tue, 16 Feb 2010 20:53 -0600, "Doc" <<a href="mailto:ledoktre@meanie.us">ledoktre@meanie.us</a>> wrote:<br>
><br>
>> 1) I am able to see a call come in, and it gets routed to the sample IVR<br>
>> to start. The first thing off is that when I dial from PSTN -><br>
>> Skype-In, it does not let me push any buttons. If I launch a second<br>
>> skype client, and dial the skype user on FS directly, it works fine.<br>
>> Any ideas?<br>
>><br>
><br>
> Hi Doc,<br>
><br>
> When you dial in from the PSTN the 'push button' events are present as<br>
> "in band" audio tones.<br>
> By default the sample IVR only works on "out of band" DTMF events.<br>
><br>
> This is why when you call directly from another skype client the 'push<br>
> button' events are detected because they are passed as "out of band"<br>
> signaling.<br>
><br>
><br>
> Now,<br>
> In the dial plan you can tell FS to listen for "in band" audio DTMF<br>
> tones using <action application="start_dtmf" /><br>
><br>
> However,<br>
> This currently does not work during a skype call for some reason?<br>
> <a href="http://jira.freeswitch.org/browse/MODSKYPIAX-66" target="_blank">http://jira.freeswitch.org/browse/MODSKYPIAX-66</a><br>
><br>
><br>
> A work around,<br>
> is to sign up for the "Skype SIP Beta" product where the 'push button'<br>
> events are sent to FS "out of band".<br>
><br>
> This conversion is done at the PSTN --> Skype gateway by dedicated DTMF<br>
> tone detection hardware.<br>
><br>
><br>
> Skype has either made a business decision, or a technical over sight to<br>
> pass DTMF events 'out of band' for only addition fee products.<br>
><br>
><br>
> It has also been reported in New Zealand that even the 'in band' tones<br>
> where present one day and actually filtered out the next.<br>
> This seems extreme, but either through deliberate action or a technology<br>
> change this is what was reported on one blog.<br>
><br>
> It is unclear to me if this was a technical limitation or a blunt<br>
> business decisions, but a audio sample showed the audio tones missing?<br>
><br>
><br>
> In any case you would not want to rely on 'In band' DTMF' tones when<br>
> passed through 'lossy' codecs anyway.<br>
><br>
> Best to stick with 'out of band' signaling for reliability.<br>
><br>
><br>
> regards,<br>
> Scott Torr<br>
><br>
><br>
><br>
><br>
><br>
><br>
><br>
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</div></div></blockquote></div><br></div>