Problem with pjsip dial


#1

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 ?


#2

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.


#3

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


#4

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


#5

SImply replace sip by pjsip in my reply.