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Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

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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
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3
apps/app_dial.c
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@@ -1427,6 +1427,9 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
cc_frame_received = 1;
}
break;
case AST_CONTROL_PVT_CAUSE_CODE:
ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
break;
case -1:
if (single && !caller_entertained) {
ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));