[asterisk-dev] Detailed SIP disconnect reason

[CDR]

I converted my dialplan to Trunk, but all my calls are being rejected with
"not acceptable here". It seems like since there is no matching peer, it
does not apply the global codec list. Any idea?
Philip

On Wed, Feb 24, 2010 at 2:23 AM, Olle E. Johansson <oej at edvina.net> wrote:

>
> 24 feb 2010 kl. 08.00 skrev Kirill 'Big K' Katsnelson:
>
> > On 100223 2245, moi self wrote:
> >> I previously was under an impression that CHANNEL() gives access to
> >> other channel variables;
> >
> > s/variables/information bits/
> >
> > Sorry for the confusion introduced.
> In trunk there's a variable called SIP_CAUSE that is a hash of all the
> responses. Doesn't that give you what you need?
>
> /O
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