9 files changed, 144 insertions, 12 deletions

diff --git a/net/asterisk/Makefile b/net/asterisk/Makefile
index 2556b4ee3aec..1b41a4d3a73d 100644
--- a/net/asterisk/Makefile
+++ b/net/asterisk/Makefile
@@ -7,11 +7,10 @@
 
 PORTNAME=	asterisk
 PORTVERSION=	1.4.24.1
+PORTREVISION=	1
 CATEGORIES=	net
-MASTER_SITES=	http://downloads.digium.com/pub/asterisk/ \
-		http://downloads.digium.com/pub/asterisk/old-releases/ \
-		http://ftp.digium.com/pub/asterisk/ \
-		http://ftp.digium.com/pub/asterisk/old-releases/
+MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/ \
+		http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
 
 MAINTAINER=	sobomax@FreeBSD.org
 COMMENT=	An Open Source PBX and telephony toolkit
diff --git a/net/asterisk/files/patch-hangup_hang b/net/asterisk/files/patch-hangup_hang
new file mode 100644
index 000000000000..22e09d6a1be7
--- /dev/null
+++ b/net/asterisk/files/patch-hangup_hang
@@ -0,0 +1,20 @@
+Index: main/channel.c
+===================================================================
+--- main/channel.c	(revision 188244)
++++ main/channel.c	(working copy)
+@@ -2038,13 +2038,13 @@
+ 			return &ast_null_frame;
+ 		usleep(1);
+ 	}
+-
++/*
+ 	if (chan->fdno == -1) {
+ 		ast_log(LOG_ERROR, "ast_read() called with no recorded file descriptor.\n");
+ 		f = &ast_null_frame;
+ 		goto done;
+ 	}
+-
++*/
+ 	if (chan->masq) {
+ 		if (ast_do_masquerade(chan))
+ 			ast_log(LOG_WARNING, "Failed to perform masquerade\n");
diff --git a/net/asterisk/files/patch-rtp_timeout b/net/asterisk/files/patch-rtp_timeout
new file mode 100644
index 000000000000..6f60c74cff41
--- /dev/null
+++ b/net/asterisk/files/patch-rtp_timeout
@@ -0,0 +1,25 @@
+--- channels/chan_sip.c.orig	2009-04-14 08:58:13.000000000 +0100
++++ channels/chan_sip.c	2009-04-14 09:00:58.000000000 +0100
+@@ -16287,7 +16287,6 @@
+ restartsearch:		
+ 		/* Check for interfaces needing to be killed */
+ 		ast_mutex_lock(&iflock);
+-		t = time(NULL);
+ 		/* don't scan the interface list if it hasn't been a reasonable period
+ 		   of time since the last time we did it (when MWI is being sent, we can
+ 		   get back to this point every millisecond or less)
+@@ -16306,12 +16305,14 @@
+ 			    (sip->owner->_state == AST_STATE_UP) &&
+ 			    !sip->redirip.sin_addr.s_addr &&
+ 			    sip->t38.state != T38_ENABLED) {
++				t = time(NULL);
+ 				if (sip->lastrtptx &&
+ 				    ast_rtp_get_rtpkeepalive(sip->rtp) &&
+ 				    (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {
+ 					/* Need to send an empty RTP packet */
+ 					sip->lastrtptx = time(NULL);
+ 					ast_rtp_sendcng(sip->rtp, 0);
++					t = time(NULL);
+ 				}
+ 				if (sip->lastrtprx &&
+ 					(ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) &&
diff --git a/net/asterisk10/Makefile b/net/asterisk10/Makefile
index 2556b4ee3aec..1b41a4d3a73d 100644
--- a/net/asterisk10/Makefile
+++ b/net/asterisk10/Makefile
@@ -7,11 +7,10 @@
 
 PORTNAME=	asterisk
 PORTVERSION=	1.4.24.1
+PORTREVISION=	1
 CATEGORIES=	net
-MASTER_SITES=	http://downloads.digium.com/pub/asterisk/ \
-		http://downloads.digium.com/pub/asterisk/old-releases/ \
-		http://ftp.digium.com/pub/asterisk/ \
-		http://ftp.digium.com/pub/asterisk/old-releases/
+MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/ \
+		http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
 
 MAINTAINER=	sobomax@FreeBSD.org
 COMMENT=	An Open Source PBX and telephony toolkit
diff --git a/net/asterisk10/files/patch-hangup_hang b/net/asterisk10/files/patch-hangup_hang
new file mode 100644
index 000000000000..22e09d6a1be7
--- /dev/null
+++ b/net/asterisk10/files/patch-hangup_hang
@@ -0,0 +1,20 @@
+Index: main/channel.c
+===================================================================
+--- main/channel.c	(revision 188244)
++++ main/channel.c	(working copy)
+@@ -2038,13 +2038,13 @@
+ 			return &ast_null_frame;
+ 		usleep(1);
+ 	}
+-
++/*
+ 	if (chan->fdno == -1) {
+ 		ast_log(LOG_ERROR, "ast_read() called with no recorded file descriptor.\n");
+ 		f = &ast_null_frame;
+ 		goto done;
+ 	}
+-
++*/
+ 	if (chan->masq) {
+ 		if (ast_do_masquerade(chan))
+ 			ast_log(LOG_WARNING, "Failed to perform masquerade\n");
diff --git a/net/asterisk10/files/patch-rtp_timeout b/net/asterisk10/files/patch-rtp_timeout
new file mode 100644
index 000000000000..6f60c74cff41
--- /dev/null
+++ b/net/asterisk10/files/patch-rtp_timeout
@@ -0,0 +1,25 @@
+--- channels/chan_sip.c.orig	2009-04-14 08:58:13.000000000 +0100
++++ channels/chan_sip.c	2009-04-14 09:00:58.000000000 +0100
+@@ -16287,7 +16287,6 @@
+ restartsearch:		
+ 		/* Check for interfaces needing to be killed */
+ 		ast_mutex_lock(&iflock);
+-		t = time(NULL);
+ 		/* don't scan the interface list if it hasn't been a reasonable period
+ 		   of time since the last time we did it (when MWI is being sent, we can
+ 		   get back to this point every millisecond or less)
+@@ -16306,12 +16305,14 @@
+ 			    (sip->owner->_state == AST_STATE_UP) &&
+ 			    !sip->redirip.sin_addr.s_addr &&
+ 			    sip->t38.state != T38_ENABLED) {
++				t = time(NULL);
+ 				if (sip->lastrtptx &&
+ 				    ast_rtp_get_rtpkeepalive(sip->rtp) &&
+ 				    (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {
+ 					/* Need to send an empty RTP packet */
+ 					sip->lastrtptx = time(NULL);
+ 					ast_rtp_sendcng(sip->rtp, 0);
++					t = time(NULL);
+ 				}
+ 				if (sip->lastrtprx &&
+ 					(ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) &&
diff --git a/net/asterisk14/Makefile b/net/asterisk14/Makefile
index 2556b4ee3aec..1b41a4d3a73d 100644
--- a/net/asterisk14/Makefile
+++ b/net/asterisk14/Makefile
@@ -7,11 +7,10 @@
 
 PORTNAME=	asterisk
 PORTVERSION=	1.4.24.1
+PORTREVISION=	1
 CATEGORIES=	net
-MASTER_SITES=	http://downloads.digium.com/pub/asterisk/ \
-		http://downloads.digium.com/pub/asterisk/old-releases/ \
-		http://ftp.digium.com/pub/asterisk/ \
-		http://ftp.digium.com/pub/asterisk/old-releases/
+MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/ \
+		http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
 
 MAINTAINER=	sobomax@FreeBSD.org
 COMMENT=	An Open Source PBX and telephony toolkit
diff --git a/net/asterisk14/files/patch-hangup_hang b/net/asterisk14/files/patch-hangup_hang
new file mode 100644
index 000000000000..22e09d6a1be7
--- /dev/null
+++ b/net/asterisk14/files/patch-hangup_hang
@@ -0,0 +1,20 @@
+Index: main/channel.c
+===================================================================
+--- main/channel.c	(revision 188244)
++++ main/channel.c	(working copy)
+@@ -2038,13 +2038,13 @@
+ 			return &ast_null_frame;
+ 		usleep(1);
+ 	}
+-
++/*
+ 	if (chan->fdno == -1) {
+ 		ast_log(LOG_ERROR, "ast_read() called with no recorded file descriptor.\n");
+ 		f = &ast_null_frame;
+ 		goto done;
+ 	}
+-
++*/
+ 	if (chan->masq) {
+ 		if (ast_do_masquerade(chan))
+ 			ast_log(LOG_WARNING, "Failed to perform masquerade\n");
diff --git a/net/asterisk14/files/patch-rtp_timeout b/net/asterisk14/files/patch-rtp_timeout
new file mode 100644
index 000000000000..6f60c74cff41
--- /dev/null
+++ b/net/asterisk14/files/patch-rtp_timeout
@@ -0,0 +1,25 @@
+--- channels/chan_sip.c.orig	2009-04-14 08:58:13.000000000 +0100
++++ channels/chan_sip.c	2009-04-14 09:00:58.000000000 +0100
+@@ -16287,7 +16287,6 @@
+ restartsearch:		
+ 		/* Check for interfaces needing to be killed */
+ 		ast_mutex_lock(&iflock);
+-		t = time(NULL);
+ 		/* don't scan the interface list if it hasn't been a reasonable period
+ 		   of time since the last time we did it (when MWI is being sent, we can
+ 		   get back to this point every millisecond or less)
+@@ -16306,12 +16305,14 @@
+ 			    (sip->owner->_state == AST_STATE_UP) &&
+ 			    !sip->redirip.sin_addr.s_addr &&
+ 			    sip->t38.state != T38_ENABLED) {
++				t = time(NULL);
+ 				if (sip->lastrtptx &&
+ 				    ast_rtp_get_rtpkeepalive(sip->rtp) &&
+ 				    (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {
+ 					/* Need to send an empty RTP packet */
+ 					sip->lastrtptx = time(NULL);
+ 					ast_rtp_sendcng(sip->rtp, 0);
++					t = time(NULL);
+ 				}
+ 				if (sip->lastrtprx &&
+ 					(ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) &&