Asterisk - The Open Source Telephony Project  18.5.0
Hangup Causes for Asterisk

The Asterisk hangup causes are delivered to the dialplan in the ${HANGUPCAUSE} channel variable after a call (after execution of "dial").

In SIP, we have a conversion table to convert between SIP return codes and Q.931 both ways. This is to improve SIP/ISDN compatibility.

These are the current codes, based on the Q.850/Q.931 specification:

For more information: