[asterisk-commits] kpfleming: branch 1.4 r46937 - /branches/1.4/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: kpfleming
Date: Thu Nov  2 10:45:32 2006
New Revision: 46937

URL: http://svn.digium.com/view/asterisk?rev=46937&view=rev
Log:
don't send INVITE when we have determined that we can't offer any audio formats due to lack of trancoding support (or incorrect configuration)

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=46937&r1=46936&r2=46937&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Nov  2 10:45:32 2006
@@ -6034,6 +6034,12 @@
 
 	/* Ok, let's start working with codec selection here */
 	capability = ast_translate_available_formats(p->jointcapability, p->prefcodec);
+
+	/* If there are no audio formats left to offer, punt */
+	if (!(capability & AST_FORMAT_AUDIO_MASK)) {
+		ast_log(LOG_WARNING, "No audio format found to offer.\n");
+		return -1;
+	}
 
 	if (option_debug > 1) {
 		char codecbuf[BUFSIZ];