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
