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 <[EMAIL PROTECTED]> Candela Technologies Inc http://www.candelatech.com
_______________________________________________ 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
