My conf files are the simplest possible:
context=default ; Default context for incoming calls
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=incoming ; the internal context controls what we can do
exten => s,1,Answer( )
exten => s,2,Playback(/var/lib/asterisk/sounds/hello-world)
exten => s,3,Hangup( )
In the documentation I read, the fact that the ‘s’ extention intercept every entering action in the context is mentionned for the FXS and FXO channels.
More precisely my questions are:
Is there an extention that match every dialed number for the SIP channels?
If yes is it the ‘s’ extetion or what?
Is there someone who succeeded in doing this task: intercepting any dialed number from a sip channel?