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Version 0.1.6 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2000-12-28 19:15:51 +00:00
parent 67826ab9b8
commit 4ea6a05122
1 changed files with 15 additions and 4 deletions
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19
apps/app_dial.c
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@@ -514,6 +514,11 @@ static int park_exec(struct ast_channel *chan, void *data)
nchan = chan;
else
nchan = ast_translator_create(chan, peer->format, AST_DIRECTION_BOTH);
if (!nchan) {
ast_log(LOG_WARNING, "Had to drop call because there was no translator for %s to %s.\n", chan->name, peer->name);
ast_hangup(peer);
return -1;
}
/* This runs sorta backwards, since we give the incoming channel control, as if it
were the person called. */
if (option_verbose > 2)
@@ -649,10 +654,16 @@ static int dial_exec(struct ast_channel *chan, void *data)
npeer = peer;
else
npeer = ast_translator_create(peer, chan->format, AST_DIRECTION_BOTH);
res = bridge_call(chan, npeer, allowredir);
if (npeer != peer)
ast_translator_destroy(npeer);
ast_hangup(peer);
if (!npeer) {
ast_log(LOG_WARNING, "Had to drop call because there was no translator for %s to %s.\n", chan->name, peer->name);
ast_hangup(peer);
res = -1;
} else {
res = bridge_call(chan, npeer, allowredir);
if (npeer != peer)
ast_translator_destroy(npeer);
ast_hangup(peer);
}
}
out:
hanguptree(outgoing, NULL);