Hello,
I've been running asterisk in one form or another for a few years with primarily internal extensions (external extensions were IAX softphones), however we are being forced to add a number of off-site SIP based phones (linksys SPA941). At the moment I've been doing some testing with Trixbox 2.8.0.3 and everything seems like it is working with the exception that any calls placed from extensions that have an external IP are going to from-sip-external and then obviously getting the congestion/number out of service recording. I've found a few sites suggesting that you should comment out most of the "stock" lines in from-sip-external and replace it with a Goto(from-pstn,s,1). My question is: Is there anyway to make the endpoint (SPA941) "authenticate/identify" itself to asterisk when a number is dialed from the handset? As of right now all calls are "unknown peer"... Hopefully that makes sense to everyone. Thanks.
-J
Line from console when dialing (dialed number changed to 1234567 for obvious reasons):
-- Executing [99051234567@from-sip-external:1] NoOp("SIP/localhost-08ee24f0", "Received incoming SIP connection from unknown peer to 99051234567") in new stack
Member Since:
2009-03-07