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
>
>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users