Fix a logic flaw I introduced when I was testing stuff out.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9ac215491ce998c39c91156133fb336a4a9839d6..89d502559521909b71558bab4444802c2b1fdc63 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6278,7 +6278,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
                        p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
                        p->t38_maxdatagram = global_t38_maxdatagram;
                }
-               if (p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) 
+               if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) 
                                || (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
                        ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
                                ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",