Asterisk 1.8-rc3 Dial(SIP/xxx) problem

Hi all,

I’ve just started to test 1.8-rc3 and all seems to be OK, but have strange Dial behavior.

Let’s assume that I have simple dialplan:
exten => _X.,1,Dial(SIP/${EXTEN})

When dialing extension 100 and device is not registered to asterisk I have
– Called SIP/100

As I can remember in 1.6.x there was no such a problem - I got simple response - Congestion or channel is unavailable.

Is there some change in 1.8 branch - for example I must use DEVICE_STATE() or ChaIsAvailable() to determine if called peer is available or this is some kind of config error in my asterisk?

Regards,
Jarek