FYI- FreePBX updates in real time using
"apply configuration changes" (Asterisk reload) as the final step in updating 
the web page interface.


Henry L.Coleman [VoIP-PBX.ca]
-------------------------------------------------



> Reza - Asterisk Consultant<
> The only way to avoid having to restart asterisk services or avoid reloading
> .conf files, is to have real time enabled - where your sip.conf entries are
> processed from a database such as mysql - then all entries, additions,
> deletes of sip accounts are real time.
>
> The reason you *have* to "reload"  or "restart" asterisk service is that the
> contents of the sip.conf file has to be parsed into the Asterisk's built in
> database engine.
>
> Not sure if FreePBX supports Real Time.   If it does, you are in luck.    We
> don't use FreePBX.  We use plain vanilla asterisk with real time enabled if
> we want real time functionality of creating accounts, extensions, voice
> mail, etc.
>
> Best regards,
> Reza.
>
> --
> Toronto based VoIP / Asterisk Trainer,
> I.T. Consultant and Hosted PBX Solutions Provider.
> +1-647-476-2067.
> http://www.linkedin.com/in/seminar
>
>
> On Fri, Nov 6, 2009 at 12:32 PM, Martin Glazer <[email protected]>wrote:
>
>> Before restarting, do you have any actual sip*.conf files in /etc/asterisk
>>
>> you may have to create a sip.conf first in order for the new extension
>> creation to work without a restart.
>>
>> Just a  thought
>>
>> Martin
>>
>>
>> Darryl Moore wrote:
>>
>>> Hey TAUGers,
>>>
>>> OK I have Asterisk and FreePBX working really well now under the latest
>>> Ubuntu. As well I have full backups working, such that installing and
>>> restoring a completely crashed system takes only about an additional 10
>>> minutes over the time to install the OS. Yeah!
>>>
>>> I only have one finicky little detail I have not been able to resolve.
>>> When I add the first SIP extension to the system (whether via the backup
>>> system, or a new extension on the web page) I always have to restart
>>> asterisk before it will see the phone.
>>>
>>> It is like there are SIP modules that it has to load, and it only loads
>>> them on startup. When I run asterisk -r and type help there are no sip
>>> commands available. As soon as I restart asterisk (assuming a SIP
>>> extension has been added) those commands will become available.
>>>
>>> Is there any way I can force the loading of these sip modules either
>>> when a sip extension is added or immediately after installation before
>>> any extensions are added? It would just be nice not to have to restart
>>> asterisk again after the installation is supposedly complete.
>>>
>>> cheers,
>>> darryl
>>>
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>>>
>>
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