[asterisk-commits] oej: branch oej/teapot-1.8 r402903 - /team/oej/teapot-1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Wed Nov 20 03:41:33 2013
New Revision: 402903

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402903
Log:
Do not update source by default when answering.

I need to figure out why this was answered. It breaks a lot of stuff, but propably fixed something along the line.
Guess it has to be optional.

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=402903&r1=402902&r2=402903
==============================================================================
--- team/oej/teapot-1.8/channels/chan_sip.c (original)
+++ team/oej/teapot-1.8/channels/chan_sip.c Wed Nov 20 03:41:33 2013
@@ -6798,7 +6798,8 @@
 
 		ast_setstate(ast, AST_STATE_UP);
 		ast_debug(1, "SIP answering channel: %s\n", ast->name);
-		ast_rtp_instance_update_source(p->rtp);
+		/* Why are we changing source here? What's the reason? */
+		/*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);