[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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