* hint : did you searched the ml first? this has been discussed a lot, even little time ago...
however... sure, just use oob dtmf like rfc2833 or sip info dtmf... so you can use a low bitrate codec and asterisk will generate them again when going to the pstn... matteo 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
