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

[Gilles]

On Mon, 20 Dec 2010 12:39:44 -0600, "Kevin P. Fleming"
<kpfleming at digium.com> wrote:
>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.

Thanks much Kevin. I found this article helpful to have a better
understanding of what a B2BUA is compared to an SIP proxy:

www.voip-info.org/wiki/view/Asterisk+SIP+not-proxy

One advantage I see in using Asterisk even when the two end-points are
SIP, is that I end up with a single application to handle calls
between end-points (SIP, VOSP, and FXO) and provide additional
features like voice-mail, etc.

But I could use a good article/book to better understand my options,
how Asterisk is different from the alternatives (Freeswitch, openSIPS,
etc.)
www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooks&field-keywords=voip

Thank you.