[asterisk-users] Asterisk - Trixbox

Tue Feb 10 10:00:38 CST 2009

It's a local CLEC, Essex Telcom.

The burden does lie with them, but I doubt they'll fix it since if you 
provision a grandstream, it works just fine.

I have a total of 5 numbers with them.  Two are on the server that is 
experiencing issues.  Another is on a different server with no issues.  The 
remaining two aren't provisioned anywhere.  I'm going to be adding another 
number shortly.

-----
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

--------------------------------------------------
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Sent: Tuesday, February 10, 2009 9:50 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk - Trixbox

> How many accounts do you have?  If just one, then a single peer should
> be fine but they should be sending the destination exten as a DID,
> obviously they are not.
>
> I think the burden of fixing it lies with them?  What carrier is this?
>
>
>
> On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett
> <asterisk-users at ics-il.net> wrote:
>> I disabled that last number's registration and moved to a new number (to
>> test each number individually without the sip debugging from the others). 
>> I
>> waited maybe 5 minutes and I restarted Asterisk to ensure the other side 
>> was
>> done with whatever it was doing.  I called the second number (8152641125)
>> and the first number (8159911010) shows up as the peer.  Not only that, 
>> but
>> with this number, there's no compatible codecs.  I ensured that both 
>> entries
>> in sip.conf were the same other than things that needed to be different 
>> such
>> as username.  I even had that entry have allow=all.  I still get the 
>> codec
>> error.
>>
>> http://pastebin.com/f5b826d62  I highlighted the lines of interest.  34 
>> is
>> the peer issue whereas 42 is the codec issue.
>>
>>
>> -----
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>>
>>
>> --------------------------------------------------
>> From: "Steve Totaro" <stotaro at first-notification.com>
>> Sent: Tuesday, February 10, 2009 7:29 AM
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Subject: Re: [asterisk-users] Asterisk - Trixbox
>>
>>> Mike,
>>>
>>> Please explain the problem more clearly and post a pastebin that shows
>>> the problem and only the problem, not a huge SIP dump.
>>>
>>> If you could point out the line numbers where you suspect an issue.
>>>
>>> Thanks,
>>> Steve
>>>
>>> On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett 
>>> <asterisk-users at ics-il.net>
>>> wrote:
>>>> Can anyone help me determine where the problem lies and how to fix it?
>>>>
>>>>
>>>> -----
>>>> Mike Hammett
>>>> Intelligent Computing Solutions
>>>> http://www.ics-il.com
>>>>
>>>>
>>>> From: Mike Hammett
>>>> Sent: Thursday, January 15, 2009 1:00 PM
>>>> To: asterisk-users at lists.digium.com
>>>> Subject: [asterisk-users] Asterisk - Trixbox
>>>> My provider migrated from an old EOL softswitch to Trixbox.
>>>>
>>>> I have a number (8159093011) on a different server on a different
>>>> network.
>>>> It appears as though the incoming calls are trying to authenticate
>>>> against
>>>> that number, which isn't present on the box.  Could someone help me
>>>> decode
>>>> this debugging output?  I was calling 8159911010.  My server is
>>>> 208.100.1.33.  Theirs is 208.1.87.235.  I solved the s@ problem on the
>>>> other
>>>> server by adding insecure settings, but that didn't seem to solve it on
>>>> this
>>>> one.
>>>>
>>>> http://pastebin.com/f5151341f
>>>>
>>>>
>>>> -----
>>>> Mike Hammett
>>>> Intelligent Computing Solutions
>>>> http://www.ics-il.com
>>>>
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 