Hello,
I am working on a system that performs high-volume outbound calls. I need to incorporate the ability to detect SIT tones for calling efficiencies and reporting “bad” telephone numbers.
Has anyone successfully integrated the ability to detect SIT with Asterisk? Any help would be greatly appreciated.
Sincerely,
May be you should look at zapateller 
I’m trying to detect SIT signals, not send them.
There is nothing that does this in Asterisk.
What kind of trunks are you using?
With T1/E1 PRIs(and with some VOIP termination providers) you get hangup codes on the D channel that can tell you when most numbers are coded as disconnected, but it does not match up to every SIT tone call you receive.
Also, the problem with SIT tones in the USA is that their delivery is not standard across carriers I’ve noticed. Most happen within 2 seconds of call acceptance but others can wait until after upto 5 rings before playing the SIT tone and others wait upto 10 seconds or after playing a message to play the SIT tone. and some carriers or people even use a bad analog recording of the SIT tone that might not be recognized anyway.
The D channel messages are often unreliable. I usually get normal clearing messages on numbers which are definitely giving SIT tones and are bad numbers. I could only find these references to sit tone detection in Asterisk:
–>Enable tone detection with PRI
–>Add special information tone detection
which are in the asterisk change log for version 0.3.0 I haven’t found where this is documented or implemented though.
If anyone has any information on dealing with bad/disconnected numbers and SIT tones, especically in the context of using Originate and the Asterisk Manager please let me know.