[Asterisk-Users] Asterisk "no one is available to take your call"

[Greg Oliver]

OK - I can successfully make calls from SIp phone through an asterisk 
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.

The problem is that if the call is not answered within ~5 seconds, * 
gives the message "no one is available to take your call" and 
disconnects the call.  If I answer b4 the 5 seconds - everything is good.

Anywhere I need to set to get around this.

I have tried the t,T settings (even though the docs say no entry is 
forever) with no luck.

Thanks,

Greg Oliver