Asterisk Settings
Posted: Sun Jan 04, 2009 5:14 pm
Mod Delete
register string should be:jayday wrote:Hello,
Register String:
(my password here)@58.96.1.2/(my number here)
It does not seen to be connecting. As it says:
Registered Trunks: 1/2.
My other DID is working but not my exetel one.
Can you help me?
Do you have SSH access to the box you are running asterisk on?jayday wrote:How do I show the incoming log?
I missed a few steps, after you reach the console, type 'core set verbose 10', this should show detailed output.jayday wrote:I opened the console and made the call and it display's nothing different.
Display:
[root@localhost ~]# asterisk -r
Asterisk 1.4.11, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.11 currently running on localhost (pid = 28668)
localhost*CLI>
That shows before, during and after the call to the exetel number. My other DID numbers also show nothing different.
It's strange that you are getting no verbose output at alljayday wrote:I have ran that command and it outputted:I made a call and nothing changed...localhost*CLI> core set verbose 10
Verbosity was 0 and is now 10
localhost*CLI>
OK - sip set debug exetel (where Exetel is the name of the trunk that you have named Exetel).jayday wrote:Not sure if that is the right commandlocalhost*CLI> sip set debug on
Usage: sip set debug
Enables dumping of SIP packets for debugging purposes
sip set debug ip <host[:PORT]>
Enables dumping of SIP packets to and from host.
sip set debug peer <peername>
Enables dumping of SIP packets to and from host.
Require peer to be registered.
localhost*CLI>