On 12/17/2010 06:25 AM, Gilles wrote:
On Thu, 16 Dec 2010 11:54:31 +0100, Gilles<[email protected]>
wrote:
Now, I'd like to be able to call any number on the Net that is
advertised as "sip:[email protected]", 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: [email protected]
Check us out at www.digium.com & www.asterisk.org
--
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