[asterisk-commits] oej: trunk r100741 - /trunk/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Tue Jan 29 02:28:43 2008
New Revision: 100741

URL: http://svn.digium.com/view/asterisk?view=rev&rev=100741
Log:
Removing code that wasn't supposed to be there at all, only at an experimental stage
before I found another solution. Thanks Kevin, for reminding me.

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=100741&r1=100740&r2=100741
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jan 29 02:28:43 2008
@@ -14400,10 +14400,6 @@
 					manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
 						"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
 						p->owner->name, p->owner->uniqueid, "SIP", p->callid, p->fullcontact, p->peername);
-				/* Set bridged channel variable */
-				bridgepeer = ast_bridged_channel(p->owner);
-				if (bridgepeer)
-					pbx_builtin_setvar_helper(bridgepeer, "SIP_BRIDGED_CALLID", p->callid);
 			} else {	/* RE-invite */
 				ast_queue_frame(p->owner, &ast_null_frame);
 			}