[Asterisk-Users] Codec routing?

[William Lloyd]

You might be able to do this in CVS head Asterisk with the SIP_HEADER  
variables and a agi script.

Need to look in the source code.

-bill

On 25-Sep-05, at 3:48 AM, Anders Svensson wrote:

> Hi! I asked this question a couple of days ago but got no answer so  
> I try again.
>
>
>
> Is it possible to route a call in * based on used codec, meaning  
> that if a user use G723 that call is routed to siptrunk 1 and a  
> user using G.729 is routed to siptrunk 2?
>
>
>
>
>
>
>
> Regards
>
> Anders Svensson
>
>
>
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