Hi, I have studied the Freeswitch doc and can't find any info related to setting up the sip users, queue, ans voicemail as a realtime DB, like Asterisk does. Is this feature available?
If not, would it be a bit too much work to write to the XML all the times? Also, is there an Asterisk AMI-equivalent feature in Freeswitch? What is that called? Thanks for your input. Pete _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
