That voicemails and  users are too integrated, coming  from Asterisk they are completely different concepts, and it makes more sence that way especially when you have dialplans sending calls to multiple voicemails even if there is only one sip user.

Sip password should be optional if using ip authentication (cidr).

Sip profiles was very confusing, internal and external, port 5060 and port 5080, it needs a lot better explanation for newbies and people coming from Asterisk.


On Oct 25, 2016 20:47, "Brian West" <brian@freeswitch.org> wrote:
I would like to get community input on what concepts caused you the most headache and any input/ideas that would help me design some tutorials / videos that would help new users coming to FreeSWITCH.

Your feedback will be greatly appreciated.

Thanks,

--

Brian West
brian@freeswitch.org


Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com (50% Discount using code FreeSwitch50)
http://www.freeswitchcookbook.com 
(50% Discount using code FreeSwitch50)
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here! | Reddit: /r/freeswitch

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org