[asterisk-users] Redirect incoming call to SIP trunk.
Steve Totaro
stotaro at asteriskhelpdesk.com
Wed Mar 6 14:54:56 CST 2013
On Wed, Mar 6, 2013 at 3:48 PM, Administrator TOOTAI <admin at tootai.net> wrote:
> Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :
>>
>> Solved.
>
>
> Great, happy for you.
>
> What would be nice is to explain how you solve it for archives. Other people
> can run in the same problematic that yours and would be happy to see your
> way to get out of it
>
I would bet you that is exactly what he did. This list has died off
so much because you can find almost every answer in the archives now.
Thanks,
Steve Totaro
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