[Asterisk-Users] dtmf for public telephony access

[Matteo Brancaleoni]

depends on the device you're using, if are supported or not.

i feel very confortable with INFO method, since
is a sip message and can be easily debugged :)

Il gio, 2004-04-15 alle 09:45, Alessio Focardi ha scritto:
> Grazie Matteo,
> 
> I looked in wiki pages, but found nothing regarding dtmf tone
> regeneration, just the indication that inbound tones are not allowed
> over low bitrate codecs.
> 
> Would you raccomend sip info or rfc2833 as tone handling method ?
> 
> P.S.
> 
> finalmente un compatriota :)
> 
> 
> MB> * hint : did you searched the ml first?
> MB> this has been discussed a lot, even little time ago...
> 
> MB> however...
> MB> sure, just use oob dtmf like rfc2833 or sip info dtmf...
> MB> so you can use a low bitrate codec and asterisk
> MB> will generate them again when going to the pstn...
> 
> MB> matteo
> 
> 
> 
> MB> Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto:
> >> Hi,
> >> 
> >> I would like to have some remote users with sip phones over adsl
> >> connections access our asterisk pbx and make out calls, currently we
> >> are using a zaptel pri interface for outdialing.
> >> 
> >> What is the right way to manage dtmf over pstn lines and still retain
> >> low bandwith occupation ?
> >> 
> >> In other words:
> >> 
> >> if I use g729 (and sip info dtmf) for sip phones - asterisk communication
> >> will asterisk be able to regenerate real tones when going out to the
> >> pstn ?
> >> 
> >> Tnx for any help ... currently I havent got g729 licenses so I cant
> >> test it out by myself.
-- 
Matteo Brancaleoni
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