[Asterisk-Users] DTMF Talk off

[Doug Crompton]

John,

 Well I have mentioned this before but not sure if I did to you... The
only was I was able to get things working well here was to use ulaw/alaw
only on the FXO and also use INBAND. This is specified on both the PSTN
tab of the 3000 and in the FXO sip.conf for it. I did not have to do this
for the fxs, line1 spa3000 port.

The other thing that is useful is to call yourself. If you have a
cellphone, call it and listen while hitting DTMF keys in both directions.
You should hear at least 500ms or more of the tone.

Doug

On Tue, 20 Jun 2006, John Millican wrote:

> I have looked at that as a solution but haven't been able to get the dtmf to
> work reliably.  When I dial a local call i get connected okay and can
> obviously connect to the second * box with out problem, the problem comes in
> trying to get the ${EXTEN} portion of the dial string
> Dial(zap/1/<my_sip_number>wwwwww${EXTEN});
>  to the second * box.  It tends to not see the DTMF correctly for the number I
> want to call.  When I watch this on the CLI through SSH on the second box i
> see the call come in and go to the correct extension where it waits for
> digits to dial and I get sporadic results, sometimes no digits are recognized
> and sometimes 2 or 3, but never all correctly. I have tried increasing, and
> decreasing the wait in the dial string in my home * with no luck.  Any hints
> on how to get the 3000 and the * box to talk better?  If I could get this to
> work through the 3000 I would be a very happy camper as it would open up some
> possibilities that I can't do cost effectively otherwise.  I will start to
> route all local calls out the 3000 though for testing in the mean time.
> Thanks for the ideas,
> John M
> On Tuesday June 20 2006 10:16 am, Doug Crompton wrote:
> > Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
> > That is what I use here and I do not load or use zap or pri modules. I use
> > the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
> > computer. You could do the same for testing of your problem.
> >
> > Doug
> >
> > On Tue, 20 Jun 2006, John Millican wrote:
> > > Okay here goes,
> > > I guess I misunderstood Doug's question about the far end interface. I
> > > have no availability for high speed internet at my house to place a VoIP
> > > call over. So, I have a standard phone plugged into the PAP2, The PAP2
> > > plugs into the network at my house to which the asterisk box is also
> > > connected,  the asterisk box has an FXO card that has the PSTN line
> > > plugged into it, this is where the ZAP channel comes in.  when i dial a
> > > local number asterisk simple dials the number out the pstn line.  If i
> > > dial a long distance number, the * box dials a local phone number that I
> > > have through my VoIP provider which is answered by an * box that I have
> > > at a different location using a line in extensions.conf like:
> > > Dial(zap/1/<my_sip_number>wwwwww${EXTEN});
> > > this way when the second * answers the phone it get the ${EXTEN} that I
> > > actually dialed and dials it out over the cable connection.  I hope i was
> > > a little clearer this time and sorry for the confusion.
> > > John M
> >
> > ****************************
> > *  Doug Crompton	   *
> > *  Richboro, PA 18954	   *
> > *  215-431-6307		   *
> > *		  	   *
> > * doug at crompton.com        *
> > * http://www.crompton.com  *
> > ****************************
> >
> >
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)

****************************
*  Doug Crompton	   *
*  Richboro, PA 18954	   *
*  215-431-6307		   *
*		  	   *
* doug at crompton.com        *
* http://www.crompton.com  *
****************************