]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Merged revisions 116799 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Fri, 16 May 2008 20:30:24 +0000 (20:30 +0000)
committerJoshua Colp <jcolp@digium.com>
Fri, 16 May 2008 20:30:24 +0000 (20:30 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines

Check to make sure an RTP structure exists before calling ast_rtp_new_source on it.
(closes issue #12669)
Reported by: sbisker

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116800 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_skinny.c

index 39aed4fc6d2ef1ffb9e90bac3bf1ae5ccb4d0075..691fae0e8343c1254d4e92a0cc8f50d7cad6a869 100644 (file)
@@ -3763,7 +3763,9 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
        case AST_CONTROL_PROCEEDING:
                break;
        case AST_CONTROL_SRCUPDATE:
-               ast_rtp_new_source(sub->rtp);
+               if (sub->rtp) {
+                       ast_rtp_new_source(sub->rtp);
+               }
                break;
        default:
                ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);