[asterisk-users] Call sip:user at domain.com?

[Kevin P. Fleming]

On 12/17/2010 06:25 AM, Gilles wrote:
> On Thu, 16 Dec 2010 11:54:31 +0100, Gilles<codecomplete at free.fr>
> wrote:
>> Now, I'd like to be able to call any number on the Net that is
>> advertised as "sip:user at domain.com", such as those:
>
> I mean: Do I really have to first create a section in sip.conf each
> time a user needs to call a number on a new SIP server?
>
> http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net

You've missed a very important point here: you are using a *SIP* 
endpoint to call a *SIP* URI. The endpoint can do that directly, and 
doesn't need any help from Asterisk to do it. If you wanted to be able 
to restrict/control such calls, you'd need to use a SIP proxy... but 
Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent, which 
means whatever URI the endpoint sends to Asterisk terminates there, and 
Asterisk constructs an outbound URI of some form, connecting the two 
channels together.

You should probably take a step back and ask yourself what value 
Asterisk would bring being in the middle between your SIP softphones and 
some random SIP endpoint out on the Internet. Once you determine that, 
you'll know whether it's worth trying to construct a solution for this 
or not.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org