Use correct variable for text SRTP media.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b41f70ca61be6c38478226218f2d71a3be4b9cdd..ec79ce4e3a7d53b22f84f7653345e425cc8f7684 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5435,7 +5435,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
                        return -1;
                }
 
-               if (p->trtp && !p->vsrtp && setup_srtp(&p->tsrtp) < 0) {
+               if (p->trtp && !p->tsrtp && setup_srtp(&p->tsrtp) < 0) {
                        ast_log(LOG_WARNING, "SRTP text setup failed\n");
                        return -1;
                }