[Asterisk-Users] SIP registration/dialing problem.

[Ben Greear]

Scott Laird wrote:

> First, what's in your extensions.conf?  That controls the flow of calls 
> once they get into the system.  There should be a context that has 
> extensions for 1001 and 1002, and sip.conf should direct calls into that 
> extension via a 'context =' line.

Indeed, I had not changed the extensions.conf at all.  After adding
some (at least mostly correct) values, I was able to make calls between
my sip phones, as well as between a soft-phone based on VOCAL and a SIP
phone.

So, I'm quite satisfied with it now, though I have barely started to
scratch the surface of the feature set.

Thanks,
Ben

-- 
Ben Greear <greearb at candelatech.com>
Candela Technologies Inc  http://www.candelatech.com