I've recently decided to spend idle cycles while waiting for various Astlinux platform builds to complete on making the contents of asterisk/config a little more complete, a little more useful, a little more real-world...
I started looking at the possibility of taking JTodd's ISN Freenum cookbook example and updating it: http://freenum.org/cookbook/#ASTERISK_CONFIG but ran into some problems having to do with the following: we use SIP handsets in house, and those operate in the redfish-solutions.com domain... but anonymous SIP (freenum) calls coming in off the internet would also be in the same domain. I can't figure out how to separate calls in the same domain (but from different endpoints) in two different contexts. And unfortunately, doc/ doesn't cover sip.conf, nor does the Asterisk Reference Manual cover it in much detail... except for a couple of related topics (shared line appearances and dahdi integration, I think). So if anyone can contact me and work off-line on getting a reasonably useful and real-world applicable example working, I'll file a documentation defect and push for getting this checked in (as I did for a few other examples). Thanks, -Philip _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
