[Asterisk-Users] dtmf for public telephony access

[Alessio Focardi]

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.



-- 
Best regards,
 Alessio                            mailto:afoc at interconnessioni.it