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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-06 17:31:38 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-06 17:31:38 +0000
commitb50a27d67712750ba92be3f96abca04849251ac6 (patch)
treeaf13ffdb44f75a2c6e0c1271ea93307e9819db22
parent882da44d4a75f89336f1cca9a677979ef125403a (diff)
Fix a logic flaw I introduced when I was testing stuff out.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@228479 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9ac215491..89d502559 100644
--- 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 " : "",