[asterisk-commits] file: trunk r45286 - in /trunk: channels/chan_sip.c configs/sip.conf.sample

[asterisk-commits at lists.digium.com]

Author: file
Date: Mon Oct 16 15:26:56 2006
New Revision: 45286

URL: http://svn.digium.com/view/asterisk?rev=45286&view=rev
Log:
In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!

Modified:
    trunk/channels/chan_sip.c
    trunk/configs/sip.conf.sample

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=45286&r1=45285&r2=45286&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Oct 16 15:26:56 2006
@@ -535,8 +535,6 @@
 /*! \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 */
@@ -4260,8 +4258,8 @@
 	}
 	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)
+	/* Responses can not create a pvt structure so drop it */
+	if (req->method == SIP_RESPONSE)
 		return NULL;
 
 	/* Allocate new call */
@@ -9993,7 +9991,6 @@
 	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");
@@ -14977,11 +14974,6 @@
 	} 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;
@@ -15671,7 +15663,6 @@
 	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));

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=45286&r1=45285&r2=45286&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Oct 16 15:26:56 2006
@@ -131,8 +131,6 @@
 				; for Sipura and Grandstream ATAs, among others). This is
 				; contrary to the RFC3551 specification, the peer _should_
 				; be negotiating AAL2-G726-32 instead :-(
-
-;ignoreoodresponses = no        ; If no then out of dialog responses will not be ignored
 
 ;
 ; If regcontext is specified, Asterisk will dynamically create and destroy a