When I create the call file i write local like - "SIP/peer/destination"
And the other params like Context, Extension etc…
While in asterisk 1.8 everything worked fine, In asterisk 12, when the call failes, it does not go to the failed extension.
the only change i made in the call file is the dial from PJSIP - “PJSIP/destination@peer”
Any one notice that?
When the call is answered, it works fine.
the SIP/peer/destination is the way to write in the call file/
the peer is the end point set at the sip.conf
in the call file it is written - Channel: SIP/$peer/$destination.
it calls directly to the destination with the peer set in the sip conf.
there is no need for logging.
it is a simple question - did any ony use call file to call directly and uses the failed extension?
if so, does it work with PJSIP in asterisk 12 ?