<br><font size=2 face="sans-serif"><snip></font>
<br><font size=2><tt>rsenykoff@harrislogic.com wrote:<br>
> Is there any way to set Asterisk to choose what codec to allow for
a new <br>
> call based on current usage?<br>
I think there is a way. Since I'm not in the stage yet to configure my
<br>
extensions.conf on that deep level I found some clues.<br>
<br>
http://www.voip-info.org/wiki-Asterisk+variables<br>
${SIP_CODEC}: Used to set the SIP codec for a call<br>
<br>
Probably if you make the call go thru an extension which checks current
<br>
bandwidth consumption via an external program. (Something AGI) You could
<br>
make the call jump to an low/normal/high bandwidth setting by set the <br>
SIP_CODEC for the to be used codec. With a bit of magic you probably can
<br>
check the amount of free G729 licences too.<br>
<br>
<br>
Greetings,<br>
<br>
Stefan de Konink<br>
<br>
ps. The idea is neat... I'm definately going to try to work out some code.</tt></font>
<br><font size=2 face="sans-serif"></snip></font>
<br>
<br><font size=2 face="sans-serif">I think this would be a great way to
provide a high level of service without needing to always up the connection.
It would just be a nice way to handle those "peak" times (big
conference call for example).</font>
<br>
<br><font size=2 face="sans-serif">Let me know if you make any progress.
I'll take a look at that variable. I'm thinking that rather than check
against bandwidth, just provide a variable that can be configured to define
the maximum number of ulaw calls. If that number is currently maxed, then
any new calls get g729. But, once a ulaw drops off, there is then one available
slot, so the next call initiated would get ulaw.</font>
<br>
<br><font size=2 face="sans-serif">-Ron</font>
<br>