[asterisk-users] Realtime: Should I say or should I go (now) ?

JR Richardson jmr.richardson at gmail.com
Fri Dec 21 17:00:19 CST 2007


> > And a free hint: if you are going to have to do anything that
> > resembles number porting, swapping extensions, etc.--don't use
> > extensions/phone numbers as SIP usernames.  You have to regenerate
> > config files, etc.  Make your SIP usernames meaningless and use
> > func_odbc to look up what extension is tied to which device.
> >
>
> I second that emotion.
>
> I consult with a bunch of people who "rolled their own" Asterisk systems
> long ago, and when the try to virtualize their system in various ways
> they find their hands are tied.  It is indescribably confusing once the
> number in sip.conf gets disengaged from the extensions in the dialplan.
>
> I wouldn't say to make the names meaningless, though; there are
> different ways to use those names so that they have useful meaning.
> Just don't make them extension numbers; it's like the TCP/IP boundary
> between layers.  See SIP for an example of the problems such a thing can
> cause :-)
>
> B.

Within my Realtime Asterisk Cluster, I use Directory Numbers (DN) for
all sip/iax devices.  These are a 5 or 6 digit number that don't mean
a whole lot until I assign an extension to it in the dial plan.

So DN 22331 could be exten 101 or exten 1001 and can be updated or
changed to a different extension in the dial plan without having to
update the device itself, unless the CID needs to be changed.

You need very good record keeping to be successful.  Also on the phone
device, the auth name or account name may be 22331 but the display
name will be 1001.  To make this change, you need a central
provisioning server, update the config file and reboot the phone to
update the display name.

Hope this helps and doesn't confuse things.

JR
-- 
JR Richardson
Engineering for the Masses



More information about the asterisk-users mailing list