Maurizio Faccio adinet wrote: > I own a TDM2400 board, with three FXO modules and one FXS. > I'am having trouble with analog sip phones, from two different > equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), > sometimes when I am calling someone, then I press flash, and then call > someone else, both calls stay connected after I hang up.
That's because you have just completed a flash-hook based transfer of the first call to the second call. If you don't want this feature, set 'transfer=no' for the relevant channels in chan_dahdi.conf. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
