[asterisk-commits] oej: branch 1.4 r45313 - in /branches/1.4: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue Oct 17 11:22:41 2006
New Revision: 45313

URL: http://svn.digium.com/view/asterisk?rev=45313&view=rev
Log:
remove the option that was not needed, really.

Modified:
    branches/1.4/   (props changed)
    branches/1.4/channels/chan_sip.c

Propchange: branches/1.4/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=45313&r1=45312&r2=45313&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Oct 17 11:22:41 2006
@@ -540,8 +540,6 @@
 /*! \brief Codecs that we support by default: */
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
 static int noncodeccapability = AST_RTP_DTMF;
-
-static int global_ignoreoodresponses = 1;
 
 /* Object counters */
 static int suserobjs = 0;                /*!< Static users */
@@ -4270,7 +4268,7 @@
 	ast_mutex_unlock(&iflock);
 
 	/* If this is a response and we have ignoring of out of dialog responses turned on, then drop it */
-	if (req->method == SIP_RESPONSE && global_ignoreoodresponses)
+	if (req->method == SIP_RESPONSE)
 		return NULL;
 
 	/* Allocate new call */
@@ -10014,7 +10012,6 @@
 	ast_cli(fd, "  Allow subscriptions:    %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
 	ast_cli(fd, "  Allow overlap dialing:  %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
 	ast_cli(fd, "  Promsic. redir:         %s\n", ast_test_flag(&global_flags[0], SIP_PROMISCREDIR) ? "Yes" : "No");
-	ast_cli(fd, "  Drop misc responses:    %s\n", global_ignoreoodresponses ? "Yes" : "No");
 	ast_cli(fd, "  SIP domain support:     %s\n", AST_LIST_EMPTY(&domain_list) ? "No" : "Yes");
 	ast_cli(fd, "  Call to non-local dom.: %s\n", allow_external_domains ? "Yes" : "No");
 	ast_cli(fd, "  URI user is phone no:   %s\n", ast_test_flag(&global_flags[0], SIP_USEREQPHONE) ? "Yes" : "No");
@@ -15034,12 +15031,7 @@
 	} else if (!strcasecmp(v->name, "rfc2833compensate")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
-	} else if (!strcasecmp(v->name, "ignoreoodresponses")) {
-		if (ast_true(v->value))
-			global_ignoreoodresponses = 1;
-		else
-			global_ignoreoodresponses = 0;
-	}
+	} 
 
 	return res;
 }
@@ -15717,7 +15709,6 @@
 	ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);	/* Default for peers, users: TRUE */
 	ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP);		/* Default for peers, users: TRUE */
 	ast_set_flag(&global_flags[1], SIP_PAGE2_RTUPDATE);
-	global_ignoreoodresponses = 1;
 
 	/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
 	ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));