16 Apr
2008
16 Apr
'08
4:44 p.m.
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