[asterisk-commits] oej: trunk r89285 - /trunk/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Thu Nov 15 06:21:57 2007
New Revision: 89285

URL: http://svn.digium.com/view/asterisk?view=rev&rev=89285
Log:
Always relying on the responses when crossing NAT's are not a good
solution, it breaks communication.
Rizzo - you need to implement a configuration option for this 
code. It's good, but maybe should be off by default.

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=89285&r1=89284&r2=89285
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov 15 06:21:57 2007
@@ -13810,7 +13810,13 @@
 		gettag(req, "To", tag, sizeof(tag));
 		ast_string_field_set(p, theirtag, tag);
 	}
-	check_via_response(p, req);
+	/* This needs to be configurable on a channel/peer/user level,
+	   not mandatory for all communication. Sadly enough, NAT implementations
+	   are not so stable so we can always rely on these headers. 
+		Temporarily disabled, while waiting for fix.
+	   Fix assigned to Rizzo :-)
+	*/
+	/* check_via_response(p, req); */
 	if (p->relatedpeer && p->method == SIP_OPTIONS) {
 		/* We don't really care what the response is, just that it replied back. 
 		   Well, as long as it's not a 100 response...  since we might