Problem with pjsip dial

I have setup Asterisk to realize audio calls via webrtc with pjsip. But i cant dial between extensions. I have two extensions 1001 and 1002 and also i have configure extensions.conf

exten => _100[12],1,Dial(PJSIP/${EXTEN})
exten => 999,1,Answer
same => n,Echo

When I make a call to extension 999 then echo is working very well, but when i try call to another extension there is no indication about incoming call

Have someone any idea what is wrong ?

You can’t have extensions without extensions.conf.

What is the contents of sip.conf defining devices SIP/1001 and SIP/1002? What does the log show when you dial extensions 1001 or 1002.

It will be hard to make people stop calling sip devices as extension, because most of them bring that idea from other systems where an endpoint is called extension

This is not configured sip.conf it is pjsip so as I wrote i dialing by pjsip/1001 or pjsip/1002

SImply replace sip by pjsip in my reply.