[asterisk-users] DTMF Corruption Problem

[Erick Perez]

I can report that with asterisk 1.2.13, internal SIP calls work
perfectly but (in my particular case) my asterisk box cannot recognize
DTMF digits when it receives a call via our SIP provider. we are both
using rfc2833 and I have tried relaxdtmf=yes/no

when i use an internal sip extension and call somebody outside via my
sip provider, dtmf is recognized.

On 11/9/06, mail-lists <mail-lists at peachnet.com> wrote:
>
> > Also, I am not using a zaptel timer.  Could this possibly be causing
> > problems with DTMF??
> I really don't know for certain but here's what I experienced: When
> calling out asterisk gives the option to allow called numbers to
> transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th
> dial string. This would very seldom work. I could hit the '#' on the
> called phone it would say 'extension' but would always reply with 'not
> valid extension'
>
> I recently upgraded to 1.2.12 and noticed that there was no ztdummy
> running! I compiled my own zaptel installed it, loaded the modules on
> boot and now the transfer works perfectly.
>
> Also: my moh wasn't working for some reason. After I installed the
> ztdummy module it works too..
>
> I'm not sure whether the transfer issue was fixed by using the ztdummy
> module or by the asterisk issue but my point is that you should always
> have the ztdummy module installed if possible.
>
> Just my .02. Hope it helps
>
>
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-- 
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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