[Freeswitch-users] Greetings and a couple of questions
Scott Torr
scott.torr.fs at letterboxes.org
Wed Feb 17 04:08:23 PST 2010
On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre at meanie.us> wrote:
> 1) I am able to see a call come in, and it gets routed to the sample IVR
> to start. The first thing off is that when I dial from PSTN ->
> Skype-In, it does not let me push any buttons. If I launch a second
> skype client, and dial the skype user on FS directly, it works fine.
> Any ideas?
Hi Doc,
When you dial in from the PSTN the 'push button' events are present as
"in band" audio tones.
By default the sample IVR only works on "out of band" DTMF events.
This is why when you call directly from another skype client the 'push
button' events are detected because they are passed as "out of band"
signaling.
Now,
In the dial plan you can tell FS to listen for "in band" audio DTMF
tones using <action application="start_dtmf" />
However,
This currently does not work during a skype call for some reason?
http://jira.freeswitch.org/browse/MODSKYPIAX-66
A work around,
is to sign up for the "Skype SIP Beta" product where the 'push button'
events are sent to FS "out of band".
This conversion is done at the PSTN --> Skype gateway by dedicated DTMF
tone detection hardware.
Skype has either made a business decision, or a technical over sight to
pass DTMF events 'out of band' for only addition fee products.
It has also been reported in New Zealand that even the 'in band' tones
where present one day and actually filtered out the next.
This seems extreme, but either through deliberate action or a technology
change this is what was reported on one blog.
It is unclear to me if this was a technical limitation or a blunt
business decisions, but a audio sample showed the audio tones missing?
In any case you would not want to rely on 'In band' DTMF' tones when
passed through 'lossy' codecs anyway.
Best to stick with 'out of band' signaling for reliability.
regards,
Scott Torr
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