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authorKinsey Moore <kmoore@digium.com>2012-05-14 19:44:27 +0000
committerKinsey Moore <kmoore@digium.com>2012-05-14 19:44:27 +0000
commitb5a6de76fc71e3f604761885dd844975a11893ee (patch)
tree5a8f74f510374ec3768b4e778a797b529133cb8a /apps/app_dial.c
parentfef9a32fb410b47da4cb195f0af2f63f8cac0f03 (diff)
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
Diffstat (limited to 'apps/app_dial.c')
-rw-r--r--apps/app_dial.c3
1 files changed, 3 insertions, 0 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c
index a5eb03675..e6d9c5448 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -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));