On Tue, Jul 31, 2007 at 10:06:30AM +0200, Jack wrote: > Tzafrir Cohen schrieb: > > On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
> >> signalling=pri_cpe > >> channel => 29-31 > >> > > > > and then in extensions.conf: > > > > [hangup-calls] > > ; not sure that this is precisly the right thing to do: > > exten => s,1,Hangup > > > > > This is a solution I was thinking about too, but there is one major problem: > > When there is a outgoing call, asterisk takes the first available > channel , in case there are no active calls this is Zap/22 for outgoing > calls in my configuration. If there is a incoming call immediatly after > the outgoing call is hangup, asterisk (or the telco?) does not take the > first available channel - which would be Zap/1 - it takes Zap/22 > instead. So with this solution this incoming call would get lost even > when there are no other incoming calls at all. This is because you use Dial(Zap/g3) . Use Dial(Zap/G3) to make asterisk start from the last. You would then need a higher load for calls to be wasted that way. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users