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Asterisk Settings

Posted: Sun Jan 04, 2009 5:14 pm
by jayday
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Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:19 pm
by jayday
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Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:19 pm
by JasonM
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?
register string should be:

<did number>:<secret>@58.96.1.2/<DID>

EDIT: missed it by 'that' much.

Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:26 pm
by jayday
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Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:39 pm
by JasonM
Can you please access the asterisk console (type asterisk -r ) and post a log of an incoming call (mask out any personal details).

Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:44 pm
by jayday
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Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:48 pm
by JasonM
jayday wrote:How do I show the incoming log?
Do you have SSH access to the box you are running asterisk on?

Once you are at the console, you should be able to type 'asterisk -r' this will open asterisk, after which make an incoming call, and view the output.

If you do not have SSH access to the asterisk box - does it have a keyboard and monitor connected? If so - login, and access asterisk, make an incoming call, and then copy the output.

Alternatively, use PuTTY (a program you can use for remote access) - login, access the asterisk console, and provide a log of an incoming call.

Re: Asterisk Settings

Posted: Sun Jan 04, 2009 5:55 pm
by jayday
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Re: Asterisk Settings

Posted: Sun Jan 04, 2009 6:12 pm
by JasonM
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.
I missed a few steps, after you reach the console, type 'core set verbose 10', this should show detailed output.

Re: Asterisk Settings

Posted: Mon Jan 05, 2009 9:25 am
by jayday
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Re: Asterisk Settings

Posted: Mon Jan 05, 2009 9:28 am
by JasonM
jayday wrote:I have ran that command and it outputted:
localhost*CLI> core set verbose 10
Verbosity was 0 and is now 10
localhost*CLI>
I made a call and nothing changed...
It's strange that you are getting no verbose output at all

Try sip set debug on, and make an incoming call.

Re: Asterisk Settings

Posted: Mon Jan 05, 2009 9:33 am
by jayday
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Re: Asterisk Settings

Posted: Mon Jan 05, 2009 9:36 am
by JasonM
jayday wrote:Not sure if that is the right command
localhost*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>
OK - sip set debug exetel (where Exetel is the name of the trunk that you have named Exetel).

Can you confirm your ATA is configured to use this asterisk server?

Re: Asterisk Settings

Posted: Mon Jan 05, 2009 10:04 am
by jayday
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Re: Asterisk Settings

Posted: Mon Jan 05, 2009 10:08 am
by JasonM
In your settings, you specified the user context as 'sipout', this should be pstn I believe (from-pstn):

'Looking for 039012**** in from-sipout'