[asterisk-commits] mmichelson: trunk r367183 - in /trunk: channels/ include/asterisk/ main/

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Mon May 21 15:31:53 2012
New Revision: 367183

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=367183
Log:
Revert revision 367163.

This should have been committed to my team trunk-digiumphones branch
instead of trunk.


Modified:
    trunk/channels/chan_sip.c
    trunk/include/asterisk/callerid.h
    trunk/main/callerid.c

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=367183&r1=367182&r2=367183
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon May 21 15:31:53 2012
@@ -671,8 +671,7 @@
 	{ AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
 	{ AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
 	{ AST_REDIRECTING_REASON_AWAY, "away" },
-	{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"},
-	{ AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm"},
+	{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
 };
 
 
@@ -24258,8 +24257,6 @@
 	int localtransfer = 0;
 	int attendedtransfer = 0;
 	int res = 0;
-	struct ast_party_redirecting redirecting;
-	struct ast_set_party_redirecting update_redirecting;
 
 	if (req->debug) {
 		ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n",
@@ -24563,16 +24560,6 @@
 		goto handle_refer_cleanup;
 	}
 	ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);	/* Delay hangup */
-
-	/* When a call is transferred to voicemail from a Digium phone, there may be
-	 * a Diversion header present in the REFER with an appropriate reason parameter
-	 * set. We need to update the redirecting information appropriately.
-	 */
-	ast_party_redirecting_init(&redirecting);
-	memset(&update_redirecting, 0, sizeof(update_redirecting));
-	change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE);
-	ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting);
-	ast_party_redirecting_free(&redirecting);
 
 	/* Do not hold the pvt lock during the indicate and async_goto. Those functions
 	 * lock channels which will invalidate locking order if the pvt lock is held.*/

Modified: trunk/include/asterisk/callerid.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/callerid.h?view=diff&rev=367183&r1=367182&r2=367183
==============================================================================
--- trunk/include/asterisk/callerid.h (original)
+++ trunk/include/asterisk/callerid.h Mon May 21 15:31:53 2012
@@ -400,7 +400,6 @@
 	AST_REDIRECTING_REASON_OUT_OF_ORDER,
 	AST_REDIRECTING_REASON_AWAY,
 	AST_REDIRECTING_REASON_CALL_FWD_DTE,           /* This is something defined in Q.931, and no I don't know what it means */
-	AST_REDIRECTING_REASON_SEND_TO_VM,
 };
 
 /*!

Modified: trunk/main/callerid.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/callerid.c?view=diff&rev=367183&r1=367182&r2=367183
==============================================================================
--- trunk/main/callerid.c (original)
+++ trunk/main/callerid.c Mon May 21 15:31:53 2012
@@ -1203,7 +1203,6 @@
 	{ AST_REDIRECTING_REASON_OUT_OF_ORDER,   "out_of_order", "Called DTE Out-Of-Order" },
 	{ AST_REDIRECTING_REASON_AWAY,           "away",         "Callee is Away" },
 	{ AST_REDIRECTING_REASON_CALL_FWD_DTE,   "cf_dte",       "Call Forwarding By The Called DTE" },
-	{ AST_REDIRECTING_REASON_SEND_TO_VM,     "send_to_vm",   "Call is being redirected to user's voicemail"},
 /* *INDENT-ON* */
 };