[Asterisk-Users] Newbee question

[Paul Mahler]

You can easily have an incoming call ring multiple extensions. You could
also send the incoming call to an alternate extension. 

 
Paul Mahler 
mail:pmahler at signate.com

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Saturday, January 17, 2004 11:34 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Newbee question

Hi!

> > in a large/ distributed environment users move about either office to 
> > office or branch to branch can they log in and have their virtual 
> > extension routed to the one they are on?
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0000102
> 
> .. currently its not somthing that is supported by Asterisk, you may be 
> able to write you own application to support it if you need it..

I agree with the comments of JT on that bug report: You can rather easily 
arrange this going using DBGet and DBPut. Still it might be a good idea 
to create an example and present that on the wiki, even include it in the 
default extensions.conf.
Apart from that you have the option to use a phone (like the SNOM) that 
allows for multiple users. But unless you take extra provisions you'll 
still need to register with _your_ server, and not the branch's server.

Let's assume we divide our number space into physical phones and personal 
numbers:

- 1000 to 1999 for physical phones
- 2000 to 2999 for persons

Now all you need to do is write a small set of macros that

- sets my caller ID (with DBget) to 2555 when I call 2301 using phone 
1022
- offers a tiny menu where I can say "I am 2555 and can be found at 1022"
and store that with DBput. You'll have to notify the other server(s) 
using AGI about this change or - better - use a shared database.

Your voicemail, however, will not move with you, so you'll still need to 
call your "home server" to retrieve that. Also the MWI function on phone 
1022 will need extra treatment in sip.conf (or whatever channel you use) 
to adjust the mailbox= setting - ok, that's a bit ugly to fix, but maybe 
you can also do without vm while away, or use the web interface...?

Cheers, Philipp


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