[asterisk-commits] jpeeler: branch jpeeler/bug12269 r168502 - in /team/jpeeler/bug12269: ./ chan...

[SVN commits to the Asterisk project]

Author: jpeeler
Date: Mon Jan 12 12:08:43 2009
New Revision: 168502

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=168502
Log:
svnmerge merge

Modified:
    team/jpeeler/bug12269/   (props changed)
    team/jpeeler/bug12269/README
    team/jpeeler/bug12269/channels/chan_sip.c
    team/jpeeler/bug12269/configs/indications.conf.sample

Propchange: team/jpeeler/bug12269/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jan 12 12:08:43 2009
@@ -1,1 +1,1 @@
-/branches/1.4:1-168285
+/branches/1.4:1-168501

Modified: team/jpeeler/bug12269/README
URL: http://svn.digium.com/svn-view/asterisk/team/jpeeler/bug12269/README?view=diff&rev=168502&r1=168501&r2=168502
==============================================================================
--- team/jpeeler/bug12269/README (original)
+++ team/jpeeler/bug12269/README Mon Jan 12 12:08:43 2009
@@ -2,7 +2,7 @@
 by Mark Spencer <markster at digium.com>
 and the Asterisk.org developer community
 
-Copyright (C) 2001-2008 Digium, Inc.
+Copyright (C) 2001-2009 Digium, Inc.
 and other copyright holders.
 ================================================================
 
@@ -11,7 +11,7 @@
 the security information file (doc/security.txt) before you attempt 
 to configure and run an Asterisk server.
 
-* WHAT IS ASTERISK ?
+* WHAT IS ASTERISK?
   Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
 sense, middleware between Internet and telephony channels on the bottom,
 and Internet and telephony applications at the top.  For more information

Modified: team/jpeeler/bug12269/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/jpeeler/bug12269/channels/chan_sip.c?view=diff&rev=168502&r1=168501&r2=168502
==============================================================================
--- team/jpeeler/bug12269/channels/chan_sip.c (original)
+++ team/jpeeler/bug12269/channels/chan_sip.c Mon Jan 12 12:08:43 2009
@@ -11332,7 +11332,6 @@
 	unsigned int event;
 	const char *c = get_header(req, "Content-Type");
 
-	check_via(p, req);
 	/* Need to check the media/type */
 	if (!strcasecmp(c, "application/dtmf-relay") ||
 	    !strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
@@ -13581,7 +13580,6 @@
 	char *eventid = NULL;
 	char *sep;
 
-	check_via(p, req);
 	if( (sep = strchr(event, ';')) ) {	/* XXX bug here - overwriting string ? */
 		*sep++ = '\0';
 		eventid = sep;
@@ -13709,7 +13707,7 @@
 {
 	int res;
 
-	check_via(p, req);
+
 	/* XXX Should we authenticate OPTIONS? XXX */
 
 	if (p->lastinvite) {
@@ -14905,7 +14903,6 @@
 
 	int res = 0;
 
-	check_via(p, req);
 	if (ast_test_flag(req, SIP_PKT_DEBUG))
 		ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");
 
@@ -15356,7 +15353,6 @@
 static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
 {
 	if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
-		check_via(p, req);
 		if (ast_test_flag(req, SIP_PKT_DEBUG))
 			ast_verbose("Receiving message!\n");
 		receive_message(p, req);

Modified: team/jpeeler/bug12269/configs/indications.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/team/jpeeler/bug12269/configs/indications.conf.sample?view=diff&rev=168502&r1=168501&r2=168502
==============================================================================
--- team/jpeeler/bug12269/configs/indications.conf.sample (original)
+++ team/jpeeler/bug12269/configs/indications.conf.sample Mon Jan 12 12:08:43 2009
@@ -117,7 +117,7 @@
 [bg]
 ; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
 description = Bulgaria
-ringdance = 1000,4000
+ringcadence = 1000,4000
 ;
 dial = 425
 busy = 425/500,0/500