at I have in my extensions.conf file > below minus my auth info :) > > What happens now is this, if I pickup the sip phone at ext. 100 and > dial extension 101 the phone at 101 rings but when 101 answers we > can't talk between the phones it's silence.
Check: Canreinvite=$value Codecs are the same on both phones _______________________________________________ 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
