[asterisk-commits] mmichelson: branch 1.4 r77824 - /branches/1.4/channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 31 10:21:23 CDT 2007


Author: mmichelson
Date: Tue Jul 31 10:21:22 2007
New Revision: 77824

URL: http://svn.digium.com/view/asterisk?view=rev&rev=77824
Log:
This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk
servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites.

(closes issue #10274, reported by cstadlmann, patched by me with approval from file)


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=77824&r1=77823&r2=77824
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jul 31 10:21:22 2007
@@ -13696,6 +13696,8 @@
 		case AST_STATE_UP:
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG, "%s: This call is UP.... \n", c->name);
+
+			transmit_response(p, "100 Trying", req);
 
 			if (p->t38.state == T38_PEER_REINVITE) {
 				struct ast_channel *bridgepeer = NULL;




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