On Mon, 20 Dec 2010 12:39:44 -0600, "Kevin P. Fleming" <[email protected]> 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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
