[asterisk-dev] Working in useful examples... and freenum/e.164 dialing in extensions.conf.example

Philip A. Prindeville philipp_subx at redfish-solutions.com
Tue Dec 1 18:26:16 CST 2009

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:


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).



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