[asterisk-commits] file: branch 1.4 r53104 - in /branches/1.4: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu Feb  1 16:24:32 2007
New Revision: 53104

URL: http://svn.digium.com/view/asterisk?view=rev&rev=53104
Log:
Merged revisions 53103 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines

Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.

........

Modified:
    branches/1.4/   (props changed)
    branches/1.4/channels/chan_sip.c

Propchange: branches/1.4/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=53104&r1=53103&r2=53104
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Feb  1 16:24:32 2007
@@ -2831,6 +2831,7 @@
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
 		p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
+		p->jointnoncodeccapability = p->noncodeccapability;
 
 		/* If there are no audio formats left to offer, punt */
 		if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {