The VOIP provider does not actually care if you are making your calls from a simple SoftPhone, or a complete PBX. Im not going to explain all the possible combinations of connecting, but i guess your confusion comes because you still dont get to the part where some one explains to you what "Native Transfer" means. IAX is a very nice protocol working behind firewalls, so, unless you configure Asterisk properly, asterisk is going to make the initial connection from your softphone to the VoIP provider, and then will transfer the call directly, so youre phone and the provider talk without intervention of Asterisk. However thats not a good thing if you want to to some billing (because Asterisk wont realize when the call ends), so in iax.conf you can configure the phone with "notransfer=yes" (please check the name of the parameter, im not sure) so Asterisk will stay in the middle of the call.
Best Regards On 4/21/06, T. Shaw <[EMAIL PROTECTED]> wrote: > Hey guys, > I'm actively trying to get the "big" picture on how all this works and > relates to each other. > I've gone through some basic examples from the book and from the sample > files just fine. > Now, I've setup an account with a VOIP provider which does IAX termination > (exgn.net) > > After getting an account and following their steps, I can make calls out > using my IAX (cubix) and Sip (Xlite) phones. > However, I'm a bit confused on the purpose on how my box asterisk box is > involved. I completely turned off my Asterisk box, and made a call out using > either of my softphones and I was successful. So I gathered that the entire > point of "iax termination" is solely for INBOUND calls TO ME (such if I have > a DID). Otherwise I'm just using them as a proxy to forward my sip traffic > to them directly from my desktop. > > I got confused because all references I have seen regarding "iax > termination" and such involved editing your local asterisk box configs as > well as the client, but really no clear mention that your config changes > only apply to INBOUND calls, and not needed if you want to just make > OUTBOUND > Sip calls. I want to do BOTH eventually, but since I still have this > learning curve, it was just another stumble for me. > > Do I have the correct picture now? > Thanks! > > Terrelle Shaw > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
