[asterisk-users] Passing DTMF

Sam asterisk at net153.net
Fri Jan 23 20:07:21 CST 2009


 From what I have read most dtmf problems are the phones them selfs.  I 
use a Grandstream HandyTone 286 ATA.  It has known dtmf isues.  However 
I have had good luck with setting both the ATA and asterisk to dtmf mode 
rfc2833.  However I would get the occasional "dtmf talk off" problem 
where people's voices would generate a dtmf tone.  A know problem with 
most ATA's.

To experiment I set the ATA to use inband dtmf and I left asterisk set 
to rfc2833.  Before this when I would call a POTS line and press a 
button on the asterisk phone I would just hear a slight blip of dtmf on 
the POTS phone.

Now since changing the ATA to inband and leaving asterisk at rfc2833, 
the dtmf going through on the POTS phone is a long tone.  I am guessing 
that since asterisk is only set to use rfc2833 in my conf, that the 
inband dtmf is passing straight through and not getting regenerated.  I 
cannot confirm yet if it has fixed my dtmf talk off problems, but I have 
not had any problems navigating through company ivr's (of course I 
didn't before either.)

Sam

Christopher Gray wrote:
> Hello:
> 
> I need to be able to reliably send out touchtone to any calling party who comes 
> into my pbx.  The standard things to help with this have been done as far as I 
> know:
> 
> 1.  dtmfmode is rfc2833.
> 
> 2.  The phones themselves are set to rfc2833.
> 
> 3.  allow=ulaw
> 
> 4.  On internal calls between extensions, touchtone works fine.
> 
> Also, I have reviewed sip.conf with my carriers.
> 
> Now for the question:  does anybody know of a carrier that can reliably allow an 
> extension in my pbx to send touchtone to a calling party?
> 
> I have tried Vitelity and VoicePulse.  Neither can do this, and VoicePulse 
> indicates they know it's a problem and will fix it at some unknown time in the 
> future.
> 
> For the curious, here is the reason for the need.  My wife, who works as a 
> translator, will use this extension to receive calls from companies needing 
> translation.  When she receives such a call, step 1 for her is to enter an 
> employee id code.  At the end of the call, she must enter an additional code to 
> receive an ending time.
> 
> Vitelity can't do this at all.  VoicePulse works about 75% of the time which is 
> not acceptable.
> 
> Thanks for any advice.
> 
> Chris
> 
> 
> 
> 
> 
>   ----------------------------------------
> Christopher Gray, President
> Bay Area Digital
> 
> Promoting good health with innovative technology
> 
> 870 Market Street, #653
> San Francisco, CA 94102
> Phone:  (415) 217-6667
> fax:    (415) 962-2520
> Email:  chris at bayareadigital.us
> 
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