[Asterisk-Users] DTMF Talk off

John Millican john at millican.us
Tue Jun 20 11:23:09 MST 2006


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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