[asterisk-commits] trunk r18797 - /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Mon Apr 10 08:19:36 2006
New Revision: 18797

URL: http://svn.digium.com/view/asterisk?rev=18797&view=rev
Log:
Small fix

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=18797&r1=18796&r2=18797&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Apr 10 08:19:36 2006
@@ -4730,7 +4730,7 @@
 	if (sipdebug)
 		add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
 	if (recordhistory)
-		append_history(p, "%s", "Re-invite sent");
+		append_history(p, "ReInv", "Re-invite sent");
 	add_sdp(&req, p);
 	/* Use this as the basis */
 	copy_request(&p->initreq, &req);
@@ -10151,6 +10151,9 @@
 	} else {	
 		/* Responses to OUTGOING SIP requests on INCOMING calls 
 		   get handled here. As well as out-of-call message responses */
+		if (ast_test_flag(req, SIP_PKT_DEBUG))
+			ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
+
 		if (resp == 200) {
 			/* Tags in early session is replaced by the tag in 200 OK, which is 
 		  	the final reply to our INVITE */