Mon Sep 20 2010 00:34:15

Asterisk developer's documentation


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.931 specification:

The range 128-255 is private cause codes. Our private causes are:

For more information: