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Problem with AudioCodes MP-118 FXO and Asterisk. Why doesn't the gateway terminate the call?
Asterisk works in tandem with AudioCodes MP-118 FXO, 4 lines are connected. The gateway is connected to Asterisk via the SIP protocol. Everything works, but there is a problem: when a call comes to the gateway and goes to Asterisk, and after the caller hangs up, the gateway does not end the call and continues to "terrorize" Asterisk. How to solve this problem?
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There must be Busy Tones parameters somewhere, such as the frequency and duration of the signal, in order for the FXO to correctly detect the busy line.
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