[asterisk-commits] file: trunk r65840 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu May 24 09:43:49 2007
New Revision: 65840

URL: http://svn.digium.com/view/asterisk?view=rev&rev=65840
Log:
Merged revisions 65839 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines

Merged revisions 65837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines

Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)

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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?view=diff&rev=65840&r1=65839&r2=65840
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu May 24 09:43:49 2007
@@ -3069,6 +3069,7 @@
 		dialog->noncodeccapability |= AST_RTP_DTMF;
 	else
 		dialog->noncodeccapability &= ~AST_RTP_DTMF;
+	dialog->jointnoncodeccapability = dialog->noncodeccapability;
 	ast_string_field_set(dialog, context, peer->context);
 	dialog->rtptimeout = peer->rtptimeout;
 	if (peer->call_limit)