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

Klaus Darilion klaus.mailinglists at pernau.at
Wed Dec 16 08:22:49 CST 2009

Philip A. Prindeville schrieb:
> 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.

Are you using different contexts? Usually, for your local attached SIP 
handsets, you have in sip.conf one peer/friend definition per SIP 
device, and use the context= option to handle calls from these devices.

Further, the sip.conf has a context defined in the general section - 
this one will be used for the unauthenticated incoming ENUM calls:

e.g. sip.conf:

then in extensions.conf you have (pseudo-code):

exten => 00........Dial(DAHDI.....)

exten => 001,1,Dial(SIP/ext001)
exten => 002,1,Dial(SIP/ext002)

include => toPstn
include => toLocalSipPhones

include => toLocalSipPhones


> 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
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