With the help of one of the providers we terminate on, I've found the
source of the problem of getting busy even when the called isn't really
busy in the absence of ANI codes in sip headers generated by asterisk.
If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can
see it holds the value '0', but seems that value won't find the way to
the sip header.
Is this an error for asterisk to not put the code or a misconfiguration
of the remote switches to drop calls without it ?
(Have I to open a bug or to request a feature ?)
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