From 40991141899da660e39cb3ab8c35b3e56b008602 Mon Sep 17 00:00:00 2001 From: Automerge script Date: Tue, 11 Apr 2006 21:06:53 +0000 Subject: [PATCH] automerge commit git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@19344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- apps/app_dial.c | 17 ++++++++++++++++- include/asterisk/linkedlists.h | 1 + 2 files changed, 17 insertions(+), 1 deletion(-) diff --git a/apps/app_dial.c b/apps/app_dial.c index 5ec6060622..c323480b26 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -1,7 +1,7 @@ /* * Asterisk -- An open source telephony toolkit. * - * Copyright (C) 1999 - 2005, Digium, Inc. + * Copyright (C) 1999 - 2006, Digium, Inc. * * Mark Spencer * @@ -827,6 +827,21 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags if (!timelimit) { timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0; warning_sound = NULL; + } else if (play_warning > timelimit) { + /* If the first warning is requested _after_ the entire call would end, + and no warning frequency is requested, then turn off the warning. If + a warning frequency is requested, reduce the 'first warning' time by + that frequency until it falls within the call's total time limit. + */ + + if (!warning_freq) { + play_warning = 0; + } else { + while (play_warning > timelimit) + play_warning -= warning_freq; + if (play_warning < 1) + play_warning = warning_freq = 0; + } } var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER"); diff --git a/include/asterisk/linkedlists.h b/include/asterisk/linkedlists.h index ed1d48340d..5f8a57b167 100644 --- a/include/asterisk/linkedlists.h +++ b/include/asterisk/linkedlists.h @@ -439,6 +439,7 @@ struct { \ if ((head)->last == (elm)) \ (head)->last = curelm; \ } \ + (elm)->field.next = NULL; \ } while (0) #endif /* _ASTERISK_LINKEDLISTS_H */