Asterisk Settings

VOIP setup and troubleshooting
jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Asterisk Settings

Post by jayday » Sun Jan 04, 2009 5:14 pm

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jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Sun Jan 04, 2009 5:19 pm

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JasonM

Re: Asterisk Settings

Post by JasonM » Sun Jan 04, 2009 5:19 pm

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.

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Sun Jan 04, 2009 5:26 pm

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JasonM

Re: Asterisk Settings

Post by JasonM » Sun Jan 04, 2009 5:39 pm

Can you please access the asterisk console (type asterisk -r ) and post a log of an incoming call (mask out any personal details).

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Sun Jan 04, 2009 5:44 pm

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Last edited by jayday on Wed Feb 25, 2009 9:46 pm, edited 1 time in total.

JasonM

Re: Asterisk Settings

Post by JasonM » Sun Jan 04, 2009 5:48 pm

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.

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Sun Jan 04, 2009 5:55 pm

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Last edited by jayday on Wed Feb 25, 2009 9:46 pm, edited 1 time in total.

JasonM

Re: Asterisk Settings

Post by JasonM » Sun Jan 04, 2009 6:12 pm

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.

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Mon Jan 05, 2009 9:25 am

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JasonM

Re: Asterisk Settings

Post by JasonM » Mon Jan 05, 2009 9:28 am

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.

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Mon Jan 05, 2009 9:33 am

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JasonM

Re: Asterisk Settings

Post by JasonM » Mon Jan 05, 2009 9:36 am

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?

jayday
Posts: 118
Joined: Thu Nov 06, 2008 9:10 pm

Re: Asterisk Settings

Post by jayday » Mon Jan 05, 2009 10:04 am

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Last edited by jayday on Wed Feb 25, 2009 9:46 pm, edited 1 time in total.

JasonM

Re: Asterisk Settings

Post by JasonM » Mon Jan 05, 2009 10:08 am

In your settings, you specified the user context as 'sipout', this should be pstn I believe (from-pstn):

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

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