From: Joshua Colp Date: Fri, 6 Nov 2009 17:31:38 +0000 (+0000) Subject: Fix a logic flaw I introduced when I was testing stuff out. X-Git-Tag: 1.6.0.18-rc1~6 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=8330be99ac9d2afd3c8617a4398493d544da5523;p=thirdparty%2Fasterisk.git 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 9ac215491c..89d5025595 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 " : "",