[asterisk-commits] mjordan: tag 1.8.8.0-rc5 r347861 - in /tags/1.8.8.0-rc5: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Dec 9 08:55:15 CST 2011
Author: mjordan
Date: Fri Dec 9 08:55:10 2011
New Revision: 347861
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=347861
Log:
Merged 347531
Modified:
tags/1.8.8.0-rc5/ (props changed)
tags/1.8.8.0-rc5/channels/chan_sip.c
Propchange: tags/1.8.8.0-rc5/
------------------------------------------------------------------------------
Binary property 'branch-1.6.2-merged' - no diff available.
Propchange: tags/1.8.8.0-rc5/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Fri Dec 9 08:55:10 2011
@@ -1,1 +1,1 @@
-/branches/1.8:339719,339779,340878,341088,343621,345063,345828-345829,347058
+/branches/1.8:339719,339779,340878,341088,343621,345063,345828-345829,347058,347531
Modified: tags/1.8.8.0-rc5/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.8.0-rc5/channels/chan_sip.c?view=diff&rev=347861&r1=347860&r2=347861
==============================================================================
--- tags/1.8.8.0-rc5/channels/chan_sip.c (original)
+++ tags/1.8.8.0-rc5/channels/chan_sip.c Fri Dec 9 08:55:10 2011
@@ -18332,11 +18332,18 @@
per device. I don't want incoming callers to record calls in my
pbx.
*/
- /* first, get the feature string, if it exists */
+
struct ast_call_feature *feat;
int j;
struct ast_frame f = { AST_FRAME_DTMF, };
+ if (!p->owner) { /* not a PBX call */
+ transmit_response(p, "481 Call leg/transaction does not exist", req);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ return;
+ }
+
+ /* first, get the feature string, if it exists */
ast_rdlock_call_features();
feat = ast_find_call_feature("automon");
if (!feat || ast_strlen_zero(feat->exten)) {
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