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 <[EMAIL PROTECTED]> 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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--
------------------------------------------------------------
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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