[asterisk-commits] file: branch 1.6.0 r228479 - /branches/1.6.0/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Fri Nov  6 11:31:38 2009
New Revision: 228479

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=228479
Log:
Fix a logic flaw I introduced when I was testing stuff out.

Modified:
    branches/1.6.0/channels/chan_sip.c

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=228479&r1=228478&r2=228479
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri Nov  6 11:31:38 2009
@@ -6278,7 +6278,7 @@
 			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 " : "",