asterisk users - Dec 2004 - Dynamically Choose Codec for Bandwidth Management

Is there any way to set Asterisk to choose what codec to allow for a new 
call based on current usage? In other words... be able to define a max 
number of ulaw calls, then after that only allowing g729? The idea here is 
that in general, a T-1 should be enough for our offices to have phone + 
citrix + some video (got good QoS in place already). But for usage spikes, 
user experience would be kept good if we could shift it into using g729.


-Ron

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rsenykoff@harrislogic.com wrote:
> Is there any way to set Asterisk to choose what codec to allow for a new 
> call based on current usage?
I think there is a way. Since I'm not in the stage yet to configure my 
extensions.conf on that deep level I found some clues.

http://www.voip-info.org/wiki-Asterisk+variables
${SIP_CODEC}: Used to set the SIP codec for a call

Probably if you make the call go thru an extension which checks current 
bandwidth consumption via an external program. (Something AGI) You could 
make the call jump to an low/normal/high bandwidth setting by set the 
SIP_CODEC for the to be used codec. With a bit of magic you probably can 
  check the amount of free G729 licences too.


Greetings,

Stefan de Konink

ps. The idea is neat... I'm definately going to try to work out some code.