[asterisk-commits] trunk r33295 - in /trunk: ./ apps/app_dial.c
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Fri Jun 9 11:12:47 MST 2006
Author: file
Date: Fri Jun 9 13:12:46 2006
New Revision: 33295
URL: http://svn.digium.com/view/asterisk?rev=33295&view=rev
Log:
Merged revisions 33294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r33294 | file | 2006-06-09 15:08:00 -0300 (Fri, 09 Jun 2006) | 2 lines
Handle hangup during recording of screened name (issue #7304 reported by kulldominique)
........
Modified:
trunk/ (props changed)
trunk/apps/app_dial.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?rev=33295&r1=33294&r2=33295&view=diff
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Fri Jun 9 13:12:46 2006
@@ -1017,11 +1017,20 @@
"At the tone, please say your name:"
*/
- ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
+ res = ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the privintro file */
- if( !ast_streamfile(chan, "vm-dialout", chan->language) )
- ast_waitstream(chan, "");
+ if (res == -1) {
+ /* Delete the file regardless since they hung up during recording */
+ ast_filedelete(privintro, NULL);
+ if( ast_fileexists(privintro,NULL,NULL ) > 0 )
+ ast_log(LOG_NOTICE,"privacy: ast_filedelete didn't do its job on %s\n", privintro);
+ else if (option_verbose > 2)
+ ast_verbose( VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", privintro);
+ goto out;
+ }
+ if( !ast_streamfile(chan, "vm-dialout", chan->language) )
+ ast_waitstream(chan, "");
}
}
}
@@ -1279,7 +1288,6 @@
res2 = ast_play_and_wait(peer,"priv-callpending");
if (!valid_priv_reply(&opts, res2))
res2 = 0;
-
/* priv-callpending script:
"I have a caller waiting, who introduces themselves as:"
*/
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