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automerge commit

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@19344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Automerge script
2006-04-11 21:06:53 +00:00
parent 2c897a8b8c
commit 4099114189
2 changed files with 17 additions and 1 deletions
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17
apps/app_dial.c
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@@ -1,7 +1,7 @@
/* /*
* Asterisk -- An open source telephony toolkit. * Asterisk -- An open source telephony toolkit.
* *
* Copyright (C) 1999 - 2005, Digium, Inc. * Copyright (C) 1999 - 2006, Digium, Inc.
* *
* Mark Spencer <markster@digium.com> * Mark Spencer <markster@digium.com>
* *
@@ -827,6 +827,21 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
if (!timelimit) { if (!timelimit) {
timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0; timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0;
warning_sound = NULL; 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"); var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");

1
include/asterisk/linkedlists.h
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@@ -439,6 +439,7 @@ struct { \
if ((head)->last == (elm)) \ if ((head)->last == (elm)) \
(head)->last = curelm; \ (head)->last = curelm; \
} \ } \
(elm)->field.next = NULL; \
} while (0) } while (0)
#endif /* _ASTERISK_LINKEDLISTS_H */ #endif /* _ASTERISK_LINKEDLISTS_H */