[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)) {




More information about the asterisk-commits mailing list