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 <[email protected]> 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
