[asterisk-commits] file: trunk r53105 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu Feb  1 16:26:11 2007
New Revision: 53105

URL: http://svn.digium.com/view/asterisk?view=rev&rev=53105
Log:
Merged revisions 53104 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines

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:
    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=53105&r1=53104&r2=53105
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Feb  1 16:26:11 2007
@@ -2961,7 +2961,8 @@
 
 	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)) {
 		ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);