Here's the output of 'sip show users':
*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 502 1234 internal No RFC35 501 1234 internal No RFC35
If you need more info, just let me know.
Time Bandit wrote:
I have 2 Gnet SIP phones connected on the same switch as the Asterisk box. So far, our phones authenticate with *, because when I do "sip show users", I see our 2 phones there.
When you say that you see them, does it look something like this :
501/501 172.16.1.201 D 255.255.255.255 5060 Unmonitored
or like this :
501/501 (Unspecified) D 255.255.255.255 5060 Unmonitored
If it is "(Unspecified)" then the phone are not registering.
The problem I have is this, when I try to dial the other extension, in this case 502, from 501, after a few seconds, I get a busy signal. If I check on the phone's logs, it says connection timeout.
Do you have the output from Asterisk's CLI ? that would help us help you
From a quick glance at your config, everything seems fine.
B.T.W. I'm near you as I live in Brossard
hth _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Jean-Francois Theroux Systems administrator PrivalODC 450.761.9973 http://www.privalodc.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
