[asterisk-gui] Users.conf questions / thoughts

[Steven Sokol]

Greetings List,

After playing with the framework for several days I have some thoughts
and questions to toss out.  Before I get into that, I did want to
thank Pari, Mark and everyone else at Digium who has worked on this.
It's a great start towards making Asterisk accessible to a significant
number of people who would otherwise run screaming to the hills.

Questions:

1) Users.conf allows you to grant a user IAX or SIP but can you also
set specific IAX or SIP options (such as auth, dtmfmode, nat, qualify)
in users.conf?

2) Is there a way to store channel variable values in users.conf (i.e.
values that are generally presented in the channel configuration files
as 'setvar=MYVAR=myvalue')?

3) Is there any fixed limit on the manager's http server that would
prevent me from building an end-user portal (i.e. a page which allows
end users to control their feature settings)?  For example, is there a
limit on simultaneous sessions?

4) I have created a user with "CTI" checked in the advanced box.
However, I don't seem to be able to log into the GUI with that user's
credentials.  The users.conf file for the user shows "hasmanager =
yes".  Is this by design or is this a bug?

5) Has Digium given any thought to Internationalization/Localization?
It looks like all of the prompts/text are embedded in the GUI at this
point.  What's a good, scalable, portable, light-weight means of
making things more portable?

Comments:

1) It seems to only really like to operate in FireFox (and other
Gecko-based browsers).  It seems to fail on IE6 and on Safari.

2) The whole Astman object could use some documentation.  I've felt my
way through creating a working screen and the associated configs, but
it took me quite a while.  If somebody (Pari?) could spare some time
to chat I would be happy to write up some basic docs.

3) I see support for Service Providers (i.e. Trunks) but I don't see
any place to add DIDs.  In fact, the Service Providers tab
specifically mentions outgoing traffic but says nothing about
incoming.  I'm working up a DIDs tab to which incoming calls can be
delivered.  Here's my thoughts:

A) DID numbers can be mapped to: Users, Menus, Queues, Mailboxes.

B) The DID context also includes a "Default" route for calls which
arrive without a DID (i.e. calls on analog lines).

C) The DID context includes the "default" context (or which ever
context you define as the destination for users).

Does this sound like a resonable plan?

4) I think the GUI could be made even better by the use of some very
simple Wizards.  The people who need the GUI are not the sort who
want/need the infinite flexibility of raw Asterisk.  If we give them a
means of adding users that asks a few very simple questions (name,
email, choice of extension number, class of service) then populates
the basics, we can eliminate a huge number of support calls.

5) Recording of menus should be built into the Voice Menus tab.

6) Backup appears to be incomplete.  At least mine did not actually
generate a tar file.  I also don't see a means of doing an automated
restore.  If the appliance supports a cron, we should be able to
schedule an auto-backup.  Better still: configure an auto-backup (if
anything has changed) which sends the configuration file to an
FTP/TFTP/DAV/etc. server.  Or emails it.

7) A few basic reports would be nice, especially a basic dialplan
report which dumps out the current dialplan as a visual directory.
(Go one step further and make it an AJAM app which shows the status of
each extension!)

What are some of the other ideas out there?

Thanks,

Steve
-- 
Steven Sokol
CEO
Sokol & Associates, LLC

Asterisk Training:  http://www.sokol-associates.com/
AstriCon 2006: http://www.astricon.net/