[asterisk-commits] oej: branch oej/teapot-1.8 r412192 - in /team/oej/teapot-1.8: channels/ patches/

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Apr 11 07:33:33 2014
New Revision: 412192

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=412192
Log:
I change my mind

Added:
    team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff   (with props)
Modified:
    team/oej/teapot-1.8/channels/chan_sip.c

Modified: team/oej/teapot-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/chan_sip.c?view=diff&rev=412192&r1=412191&r2=412192
==============================================================================
--- team/oej/teapot-1.8/channels/chan_sip.c (original)
+++ team/oej/teapot-1.8/channels/chan_sip.c Fri Apr 11 07:33:33 2014
@@ -6832,8 +6832,7 @@
 
 		ast_setstate(ast, AST_STATE_UP);
 		ast_debug(1, "SIP answering channel: %s\n", ast->name);
-		/* Why are we changing source here? What's the reason? */
-		/*ast_rtp_instance_update_source(p->rtp); */
+		ast_rtp_instance_update_source(p->rtp); 
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		start_rtcp_events(p, sched);

Added: team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff?view=auto&rev=412192
==============================================================================
--- team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff (added)
+++ team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff Fri Apr 11 07:33:33 2014
@@ -1,0 +1,14 @@
+Index: channels/chan_sip.c
+===================================================================
+--- channels/chan_sip.c	(revision 412191)
++++ channels/chan_sip.c	(working copy)
+@@ -6832,8 +6832,7 @@
+ 
+ 		ast_setstate(ast, AST_STATE_UP);
+ 		ast_debug(1, "SIP answering channel: %s\n", ast->name);
+-		/* Why are we changing source here? What's the reason? */
+-		/*ast_rtp_instance_update_source(p->rtp); */
++		ast_rtp_instance_update_source(p->rtp); 
+ 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
+ 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+ 		start_rtcp_events(p, sched);

Propchange: team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff
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Propchange: team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff
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Propchange: team/oej/teapot-1.8/patches/reinstate-src-update-after-200ok-reinvite.diff
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