[asterisk-users] Codec choice

[Sherwood McGowan]

1. Set up a Global Variable that will store that kit's current number of calls
2. Check that variable when a call starts (but before you dial out)
3. If the number of calls is <49 (since the current call will make
50), use codec A via the CHANNEL() function, otherwise use codec B
using the same function.
4. Increment the variable
5. place call
6., upon hangup, decrement the variable

Cheers

On Fri, Aug 20, 2010 at 9:06 AM, Deepika Nijhawan
<deepika.nijhawan at oxygen8.com> wrote:
> Hi,
>
>
>
> Thanks. Actually can it be done on whole kit basis rather than for an
> extension or peer.  Like if there are lot of inbound sip interconnects on a
> kit , how can we send first 50% simultaneous calls to dahdi with codec A and
> after that with codec B.
>
>
>
> Thanks,
>
> D
>
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