Soooooooooooooooooooooooooo, After talking to a few people over the last few days, it's clear to me, we each have our own way / vision how an ARI app_dial replacement would work. What I was hoping for, is we could all come together for a festivus miracle for a little bit of design work. So, lets start with the very basic. Could somebody explain like I am 5 how asterisk would do the following: [makecall] exten => 100,1,NoOp() same => n,Answer() same => n,Dial(SIP/alice@xxxxxxxxxxx) Now, I call into Asterisk, hitting 100@makecall, dial alice@xxxxxxxxxxx, far end answers, we are bridged. So, how would that look in ARI? Answer channel invoke originate (far end answers) create bridge move channel A and B into bridge Right? Some differences I can see right from the start, in the ARI example caller A does not hear ringback, with app_dial they would. So my questions are: When does app_dial indicate ringing to the channel? When does app_dial create the bridge? Does caller A live in the bridge before Caller B is answered? If you had to do this in ARI, how would you do it? Because, what I'd like to do, is come up with the spec for it, write it, then see about making a few simple demos for different languanges. I know mjordan has some examples on github, but want to get more. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger@xxxxxxxxxxxxxx | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger _______________________________________________ asterisk-app-dev mailing list asterisk-app-dev@xxxxxxxxxxxxxxxx http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev