Thank for the help :) Then i can just hope it gets fixed soon. (But now that i know about it, its not as critical anymore).
//Joakim On Mar 12, 2010, at 8:24 PM, Kevin P. Fleming wrote: > Joakim Eriksson wrote: >> I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. >> When a user calls from skype (not skype-in) to asterisk, dtmf (basically >> menus for a conference system) works just fine. >> But when a user from the inside (soft or hardware sip phone) calls out via >> skype-out dtmf doesn't work. >> I have tried setting the codec to alaw, and dtmfmode to all possible options >> (auto, inband and rfc2833). > > This is a known issue with SkypeIn and SkypeOut and is being addressed. > There should be a Skype For Asterisk release soon that contains the > changes required on its send; there are also changes being made in the > SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-) > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users