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 Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
