Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kinsey Moore
2012-05-14 19:44:27 +00:00
parent fef9a32fb4
commit b5a6de76fc
26 changed files with 101 additions and 23 deletions

View File

@@ -877,6 +877,9 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
* the caller.
*/
break;
case AST_CONTROL_PVT_CAUSE_CODE:
ast_indicate_data(caller, f->subclass.integer, f->data.ptr, f->datalen);
break;
case -1:
ast_verb(3, "%s stopped sounds\n", ast_channel_name(winner));
break;