[asterisk-users] DTMF problems

[Moises Silva]

Ok, with SIP you can send the DTMF in 3 flavors. You need to know how
your SIP telephony gateway providers send and expect the DTMF. You
configure that in Asterisk file sip.conf, look for the peer parameter
"dtmfmode", valid values are:

dtmfmode=info
Use SIP INFO messages to send, this is out of band

dtmfmode=rfc2833
Actually i dont know, but check RFC2833 :)

dtmfmode=inband
The DTMF digits are sent in the same stream that the audio. This means that
if the audio codec is of low quality, DTMF may not pass.

dtmfmode=auto
Asterisk is supposed to detect the correct DTMF mode to use, actually
I havent used this one, but you can give it a try :)


Regards

On 8/7/06, Kohler, Jeffrey <J.Kohler at techsmith.com> wrote:
> Sorry, I'm using Asterisk to dial out to the first user using SIP and a
> telephony gateway provider.  I then dial out to the second user using
> SIP and a telephony gateway provider and bridge the two calls.
>
> The first user can press keys on their telephone if necessary to be
> connected with the second user (if for example the second user is at a
> company and needs to be dialed by extension).  I guess this would be
> 'inband DTMF'.
>
> If this problem is indeed caused by poor sound quality, are there any
> suggestions for improvement?
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Moises
> Silva
> Sent: Monday, August 07, 2006 10:09 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF problems
>
> Well, since you are not providing technical information like
> protocols, I only can tell you that if you are using inband DTMF, yes,
> is possible that poor quality in the link makes DTMF go wrong.
>
> Regards
>
> On 8/7/06, Kohler, Jeffrey <J.Kohler at techsmith.com> wrote:
> >
> >
> >
> >
> > I'm having problems where touch tones are being misinterpreted.
> >
> >
> >
> > I have a setup where one user can dial out through Asterisk to second
> user's
> > telephone.
> >
> >
> >
> > (User1 - real phone - VOIP provider - Asterisk - VOIP provider - real
> phone
> > - user2)
> >
> >
> >
> > My problem occurs when you need to dial an extension or navigate a
> phone
> > tree to reach the second user.  When the first user uses their
> telephone
> > keypad to enter the extension, their keypresses are often
> misinterpreted by
> > the second user's phone system.
> >
> >
> >
> > I'm wondering if poor sound quality could cause the tones to become
> altered?
> >  Any ideas or suggestions?
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>
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