"The truth is out there" You just need a good shovel...
PD On Sun, May 15, 2005 at 04:53:58PM -0400, Jeffrey Starin wrote: > Jonathan! You don't know how much that simple explanation has helped me > understand Asterisk. Well done. Well said. And to the point clearly. > > I would hope this could find it's way onto the Asterisk Wiki and be the > *first* thing someone reads when looking at the documentation about sip. > > Thanks a million! > > J. > > Johnathan Corgan wrote: > > >Jeffrey Starin wrote: > > > >>I have a little confusion about the general settings (other than the > >>register values) in the SIP > >>General area. > > > > > >[snip] > > > >>However, I'm confused as to the purpose of the > >>"general" settings -- to what or which connection do they apply? Since > >>the context suggested for the general settings is something like > >>"nothing" to avoid unwanted sip calls, I'm confused as to the purposse > >>of values in the general section since all parameters for > >>communications with VOIP providers is contained in the contexts such > >>as I specified above, i.e., [FWD] or for example [BROADVOICE]. Can > >>someone shed some light on that for me? > > > > > >Well, your confusion is understandable, given the way sip.conf works. > > Parameters which affect incoming calls are not separated from those > >that affect outgoing calls, so it's easy to get mixed up (well, for > >me, anyway.) > > > >In the [general] section the parameters become the defaults used > >unless overridden in a specific peer section. Also, if an incoming or > >outgoing SIP call doesn't match a specific peer section, these > >parameters get used. > > > >So, for example, if you don't want any incoming SIP calls that aren't > >from a known provider, you can set the default context to something > >innocuous as you describe above and the call will get rejected as a > >non-existent context. This is what you described in our original mail. > > > >But also in this [general] section are settings for *outbound* calls > >using SIP that aren't using a specific peer section. This can be done > >with the Dial command, using a dial string such as > >IP/[EMAIL PROTECTED] where provider.com is not listed in sip.conf. > >This might happen, say, in the case of using the ENUM lookup > >capability, where the outbound SIP address of a phone number is > >determined dynamically at call time rather than pre-configured in > >extensions.conf/sip.conf. > > > >Personally, I'd like to see this changed so there are two 'general' > >sections--one for default parameters to use unless overridden when > >there *is* a peer section below, and a different one to describe > >parameters to use when the remote peer is not previously known. I > >know there are ways to accomplish this with the existing sip.conf > >structure but it seems very counter-intuitive. > > > >-Johnathan > > > >_______________________________________________ > >Asterisk-Users mailing list > >[email protected] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Pizco Dominguez -------------------------------------------------------------------------- -------------------------------------------------------------------------- GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D FINGERPRINT: 85CB 4323 F322 5837 EDB5 2033 6FB2 C326 8DE3 7A4D -------------------------------------------------------------------------- -------------------------------------------------------------------------- _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
