Merged revisions 166861 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines

Update app_queue to deal with the removal of AST_PBX_KEEPALIVE

When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.

I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.


........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@166862 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1 insertions, 2 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index d676fb5f5..c4ba19103 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -1840,8 +1840,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 						ast_pbx_run_args(peer, &args);
 					}
 					ast_free(gosub_args);
-					if (option_debug)
-						ast_log(LOG_DEBUG, "Gosub exited with status %d\n", res);
+					ast_debug(1, "Gosub exited with status %d\n", res);
 				} else {
 					ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
 				}