I've a CallManager configured with a SIP trunk to route some calls to an asterisk machine.
The problem is that, when we reach last call into asterisk, we can't make any other call.
We have configured on RouteList a second gateway (the principal Voip cisco router), so we want that when asterisk has all channels busy, CCM try to call through Router.
But this doesn't happen.
Anyone know how to solve this problem?