From b5a6de76fc71e3f604761885dd844975a11893ee Mon Sep 17 00:00:00 2001 From: Kinsey Moore Date: Mon, 14 May 2012 19:44:27 +0000 Subject: 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 --- apps/app_dial.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'apps/app_dial.c') 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)); -- cgit v1.2.3