[asterisk-commits] kpfleming: trunk r46938 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: kpfleming
Date: Thu Nov  2 10:45:50 2006
New Revision: 46938

URL: http://svn.digium.com/view/asterisk?rev=46938&view=rev
Log:
Merged revisions 46937 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006) | 2 lines

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:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=46938&r1=46937&r2=46938&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov  2 10:45:50 2006
@@ -6095,6 +6095,12 @@
 
 	/* 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];