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. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
