Remove last 1.6.2 branch ebuild that uses in-tree patching. Clean up surrounding directories accordingly.

(Portage version: 2.1.9.25/cvs/Linux x86_64)

12 files changed, 0 insertions, 1022 deletions

diff --git a/net-misc/asterisk/files/1.6.1/asterisk-1.6.1-uclibc.patch b/net-misc/asterisk/files/1.6.1/asterisk-1.6.1-uclibc.patch
deleted file mode 100644
index 36b49e60b19e..000000000000
--- a/net-misc/asterisk/files/1.6.1/asterisk-1.6.1-uclibc.patch
+++ /dev/null
@@ -1,13 +0,0 @@
---- Makefile.old	2008-03-22 11:51:39.000000000 +0100
-+++ Makefile	2008-03-22 11:51:49.000000000 +0100
-@@ -259,6 +259,10 @@
-   ASTCFLAGS+=-pthread
- endif
- 
-+ifeq ($(OSARCH),linux-uclibc)
-+  AST_LIBS+=-lpthread -ldl
-+endif
-+
- ifeq ($(OSARCH),SunOS)
-   ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2
- endif
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.11-strip-noapi.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.11-strip-noapi.patch
deleted file mode 100644
index 4861a6cdc932..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.11-strip-noapi.patch
+++ /dev/null
@@ -1,23 +0,0 @@
-diff -uNr asterisk-1.6.2.2.ORIG/build_tools/strip_nonapi asterisk-1.6.2.2/build_tools/strip_nonapi
---- asterisk-1.6.2.2.ORIG/build_tools/strip_nonapi	2008-06-19 21:26:50.000000000 +0500
-+++ asterisk-1.6.2.2/build_tools/strip_nonapi	2010-02-10 20:15:13.000000000 +0400
-@@ -29,10 +29,15 @@
- 
- case "${OSARCH}" in
-     linux-gnu|FreeBSD)
--	nm ${1} | ${GREP} -e "$TEXTSYM" | cut -d" " -f3 | ${FILTER} > striplist
--	sed -e "s/^/-N /" striplist | xargs -n 40 ${STRIP} ${1}
--	rm -f striplist
--	;;
-+		nm ${1} > symlist
-+		if [ -s symlist ]
-+		then
-+			${GREP} -e "$TEXTSYM" symlist | cut -d" " -f3 | ${FILTER} > striplist
-+			sed -e "s/^/-N /" striplist | xargs -n 40 ${STRIP} ${1}
-+			rm -f striplist
-+		fi
-+		rm -f symlist
-+		;;
-     *)
- 	;;
- esac
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-alarm-receiver-use-playtones.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-alarm-receiver-use-playtones.patch
deleted file mode 100644
index e434a12aa1e9..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-alarm-receiver-use-playtones.patch
+++ /dev/null
@@ -1,151 +0,0 @@
-diff -uNr asterisk-1.6.2.13.ORIG//apps/app_alarmreceiver.c asterisk-1.6.2.13/apps/app_alarmreceiver.c
---- asterisk-1.6.2.13.ORIG//apps/app_alarmreceiver.c	2010-10-15 18:17:49.000000000 +0100
-+++ asterisk-1.6.2.13/apps/app_alarmreceiver.c	2010-10-15 18:18:50.000000000 +0100
-@@ -51,6 +51,7 @@
- #include "asterisk/callerid.h"
- #include "asterisk/astdb.h"
- #include "asterisk/utils.h"
-+#include "asterisk/indications.h"
- 
- #define ALMRCV_CONFIG "alarmreceiver.conf"
- #define ADEMCO_CONTACT_ID "ADEMCO_CONTACT_ID"
-@@ -139,86 +140,6 @@
- 	return;
- }
- 
--
--/*
--* Build a MuLaw data block for a single frequency tone
--*/
--static void make_tone_burst(unsigned char *data, float freq, float loudness, int len, int *x)
--{
--	int     i;
--	float   val;
--
--	for (i = 0; i < len; i++) {
--		val = loudness * sin((freq * 2.0 * M_PI * (*x)++)/8000.0);
--		data[i] = AST_LIN2MU((int)val);
--	}
--
--	/* wrap back around from 8000 */
--
--	if (*x >= 8000)
--		*x = 0;
--	return;
--}
--
--/*
--* Send a single tone burst for a specifed duration and frequency.
--* Returns 0 if successful
--*/
--static int send_tone_burst(struct ast_channel *chan, float freq, int duration, int tldn)
--{
--	int res = 0;
--	int i = 0;
--	int x = 0;
--	struct ast_frame *f, wf;
--	
--	struct {
--		unsigned char offset[AST_FRIENDLY_OFFSET];
--		unsigned char buf[640];
--	} tone_block;
--
--	for (;;) {
--
--		if (ast_waitfor(chan, -1) < 0) {
--			res = -1;
--			break;
--		}
--
--		f = ast_read(chan);
--		if (!f) {
--			res = -1;
--			break;
--		}
--
--		if (f->frametype == AST_FRAME_VOICE) {
--			wf.frametype = AST_FRAME_VOICE;
--			wf.subclass = AST_FORMAT_ULAW;
--			wf.offset = AST_FRIENDLY_OFFSET;
--			wf.mallocd = 0;
--			wf.data.ptr = tone_block.buf;
--			wf.datalen = f->datalen;
--			wf.samples = wf.datalen;
--			
--			make_tone_burst(tone_block.buf, freq, (float) tldn, wf.datalen, &x);
--
--			i += wf.datalen / 8;
--			if (i > duration) {
--				ast_frfree(f);
--				break;
--			}
--			if (ast_write(chan, &wf)) {
--				ast_verb(4, "AlarmReceiver: Failed to write frame on %s\n", chan->name);
--				ast_log(LOG_WARNING, "AlarmReceiver Failed to write frame on %s\n",chan->name);
--				res = -1;
--				ast_frfree(f);
--				break;
--			}
--		}
--
--		ast_frfree(f);
--	}
--	return res;
--}
--
- /*
- * Receive a string of DTMF digits where the length of the digit string is known in advance. Do not give preferential
- * treatment to any digit value, and allow separate time out values to be specified for the first digit and all subsequent
-@@ -433,19 +354,29 @@
- 	database_increment("calls-received");
- 
- 	/* Wait for first event */
--	ast_verb(4, "AlarmReceiver: Waiting for first event from panel\n");
-+	ast_verb(4, "AlarmReceiver: Waiting for first event from panel...\n");
- 
- 	while (res >= 0) {
- 		if (got_some_digits == 0) {
- 			/* Send ACK tone sequence */
- 			ast_verb(4, "AlarmReceiver: Sending 1400Hz 100ms burst (ACK)\n");
--			res = send_tone_burst(chan, 1400.0, 100, tldn);
--			if (!res)
--				res = ast_safe_sleep(chan, 100);
-+			res = ast_playtones_start(chan, tldn, "1400", 0);
- 			if (!res) {
-+				ast_safe_sleep(chan, 100);
-+				ast_playtones_stop(chan);
-+			}
-+			if (!res) {
-+				ast_safe_sleep(chan, 100);
- 				ast_verb(4, "AlarmReceiver: Sending 2300Hz 100ms burst (ACK)\n");
--				res = send_tone_burst(chan, 2300.0, 100, tldn);
-+				res = ast_playtones_start(chan, tldn, "2300", 0);
-+				if (!res) {
-+					ast_safe_sleep(chan, 100);
-+					ast_playtones_stop(chan);
-+				}
-+			} else {
-+				ast_debug(1, "AlarmReceiver: Failed sending tones\n");
- 			}
-+
- 		}
- 		if ( res >= 0)
- 			res = receive_dtmf_digits(chan, event, sizeof(event) - 1, fdto, sdto);
-@@ -552,9 +483,14 @@
- 		if (res == 0)
- 			res = ast_safe_sleep(chan, 200);
- 
--		/* Send the kissoff tone */
--		if (res == 0)
--			res = send_tone_burst(chan, 1400.0, 900, tldn);
-+		/* Send the kissoff tone (1400 Hz, 900 ms) */
-+		if (res == 0) {
-+			res = ast_playtones_start(chan, tldn, "1400", 0);
-+			if (res == 0) {
-+			  	ast_safe_sleep(chan, 900);
-+				ast_playtones_stop(chan);
-+			}
-+		}
- 	}
- 
- 	return res;
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-backport-bri-net-ptmp.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-backport-bri-net-ptmp.patch
deleted file mode 100644
index 18f8dcce8090..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-backport-bri-net-ptmp.patch
+++ /dev/null
@@ -1,35 +0,0 @@
-diff -uNr asterisk-1.6.2.13.ORIG//channels/chan_dahdi.c asterisk-1.6.2.13/channels/chan_dahdi.c
---- asterisk-1.6.2.13.ORIG//channels/chan_dahdi.c	2010-09-16 18:25:04.000000000 +0100
-+++ asterisk-1.6.2.13/channels/chan_dahdi.c	2010-09-16 18:25:21.000000000 +0100
-@@ -16630,7 +16630,12 @@
- 					confp->chan.sig = SIG_BRI_PTMP;
- 					confp->pri.nodetype = PRI_CPE;
- 				} else if (!strcasecmp(v->value, "bri_net_ptmp")) {
-+#if defined(HAVE_PRI_CALL_HOLD)
-+					confp->chan.sig = SIG_BRI_PTMP;
-+					confp->pri.nodetype = PRI_NETWORK;
-+#else
- 					ast_log(LOG_WARNING, "How cool would it be if someone implemented this mode!  For now, sucks for you. (line %d)\n", v->lineno);
-+#endif
- 				} else if (!strcasecmp(v->value, "gr303fxoks_net")) {
- 					confp->chan.sig = SIG_GR303FXOKS;
- 					confp->pri.nodetype = PRI_NETWORK;
-diff -uNr asterisk-1.6.2.13.ORIG//configure.ac asterisk-1.6.2.13/configure.ac
---- asterisk-1.6.2.13.ORIG//configure.ac	2010-09-16 18:25:04.000000000 +0100
-+++ asterisk-1.6.2.13/configure.ac	2010-09-16 18:25:21.000000000 +0100
-@@ -334,6 +334,7 @@
- AST_EXT_LIB_SETUP([PORTAUDIO], [PortAudio], [portaudio])
- AST_EXT_LIB_SETUP([PRI], [ISDN PRI], [pri])
- AST_EXT_LIB_SETUP_DEPENDENT([PRI_INBANDDISCONNECT], [ISDN PRI set_inbanddisconnect], [PRI], [pri])
-+AST_EXT_LIB_SETUP_DEPENDENT([PRI_CALL_HOLD], [ISDN PRI call hold], [PRI], [pri])
- AST_EXT_LIB_SETUP([PWLIB], [PWlib], [pwlib])
- AST_EXT_LIB_SETUP([RADIUS], [Radius Client], [radius])
- AST_EXT_LIB_SETUP([RESAMPLE], [LIBRESAMPLE], [resample])
-@@ -1490,6 +1491,7 @@
- AST_EXT_LIB_CHECK([PORTAUDIO], [portaudio], [Pa_GetDeviceCount], [portaudio.h])
- 
- AST_EXT_LIB_CHECK([PRI], [pri], [pri_new_bri], [libpri.h])
-+AST_EXT_LIB_CHECK([PRI_CALL_HOLD], [pri], [pri_hold_enable], [libpri.h])
- 
- AST_EXT_LIB_CHECK([PRI_PROG_W_CAUSE], [pri], [pri_progress_with_cause], [libpri.h])
- 
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-confbridge-menu-invocation.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-confbridge-menu-invocation.patch
deleted file mode 100644
index 9685af91a53a..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-confbridge-menu-invocation.patch
+++ /dev/null
@@ -1,13 +0,0 @@
-Index: apps/app_confbridge.c
-===================================================================
---- apps/app_confbridge.c	(revision 271056)
-+++ apps/app_confbridge.c	(working copy)
-@@ -734,7 +734,7 @@
- 
- 	/* If the menu option is enabled provide a user or admin menu as a custom feature hook */
- 	if (ast_test_flag(&conference_bridge_user.flags, OPTION_MENU)) {
--		ast_bridge_features_hook(&conference_bridge_user.features, "#", menu_callback, &conference_bridge_user);
-+		ast_bridge_features_hook(&conference_bridge_user.features, "*", menu_callback, &conference_bridge_user);
- 	}
- 
- 	/* If the caller should be joined already muted, make it so */
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-dahdiras-without-root.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-dahdiras-without-root.patch
deleted file mode 100644
index 94f3d3681288..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-dahdiras-without-root.patch
+++ /dev/null
@@ -1,12 +0,0 @@
-diff -uNr asterisk-1.6.2.13.ORIG/apps/app_dahdiras.c asterisk-1.6.2.13/apps/app_dahdiras.c
---- asterisk-1.6.2.13.ORIG/apps/app_dahdiras.c	2010-09-16 17:40:39.000000000 +0100
-+++ asterisk-1.6.2.13/apps/app_dahdiras.c	2010-09-16 17:41:05.000000000 +0100
-@@ -119,8 +119,6 @@
- 		c = strsep(&stringp, ",");
- 	}
- 
--	argv[argc++] = "plugin";
--	argv[argc++] = "dahdi.so";
- 	argv[argc++] = "stdin";
- 
- 	/* Finally launch PPP */
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-iax2-peerstate.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-iax2-peerstate.patch
deleted file mode 100644
index c9eb8d579205..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-iax2-peerstate.patch
+++ /dev/null
@@ -1,12 +0,0 @@
-diff -uNr asterisk-1.6.2.13.ORIG/channels/chan_iax2.c asterisk-1.6.2.13/channels/chan_iax2.c
---- asterisk-1.6.2.13.ORIG/channels/chan_iax2.c	2010-09-16 13:18:19.000000000 +0100
-+++ asterisk-1.6.2.13/channels/chan_iax2.c	2010-09-16 13:18:41.000000000 +0100
-@@ -8403,7 +8403,7 @@
- 			ast_db_put("IAX/Registry", p->name, data);
- 			ast_verb(3, "Registered IAX2 '%s' (%s) at %s:%d\n", p->name,
- 					    ast_test_flag(&iaxs[callno]->state, IAX_STATE_AUTHENTICATED) ? "AUTHENTICATED" : "UNAUTHENTICATED", ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
--			manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: IAX2\r\nPeer: IAX2/%s\r\nPeerStatus: Registered\r\n", p->name);
-+			manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: IAX2\r\nPeer: IAX2/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPost: %d\r\n", p->name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
- 			register_peer_exten(p, 1);
- 			ast_devstate_changed(AST_DEVICE_UNKNOWN, "IAX2/%s", p->name); /* Activate notification */
- 		} else if (!ast_test_flag(p, IAX_TEMPONLY)) {
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-pbxstart-failed-spurious-bye.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-pbxstart-failed-spurious-bye.patch
deleted file mode 100644
index 1a47eff12cb5..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.13-pbxstart-failed-spurious-bye.patch
+++ /dev/null
@@ -1,17 +0,0 @@
----
- channels/chan_sip.c |    2 ++
- 1 file changed, 2 insertions(+)
-
-Index: b/channels/chan_sip.c
-===================================================================
---- a/channels/chan_sip.c
-+++ b/channels/chan_sip.c
-@@ -20871,6 +20871,8 @@ static int handle_request_invite(struct 
- 
- 				if (result) {
- 
-+					sip_alreadygone(p);
-+
- 					/* Unlock locks so ast_hangup can do its magic */
- 					ast_channel_unlock(c);
- 					sip_pvt_unlock(p);
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.2-nv-faxdetect.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.2-nv-faxdetect.patch
deleted file mode 100644
index ee2ef4408cf6..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.2-nv-faxdetect.patch
+++ /dev/null
@@ -1,650 +0,0 @@
-diff -uNr apps/app_nv_backgrounddetect.c apps/app_nv_backgrounddetect.c
---- apps/app_nv_backgrounddetect.c	1970-01-01 01:00:00.000000000 +0100
-+++ apps/app_nv_backgrounddetect.c	2010-02-10 22:09:40.118390386 +0000
-@@ -0,0 +1,325 @@
-+/*
-+ * Asterisk -- A telephony toolkit for Linux.
-+ *
-+ * Playback a file with audio detect
-+ * 
-+ * Copyright (C) 2004-2005, Newman Telecom, Inc. and Newman Ventures, Inc.
-+ *
-+ * Justin Newman <jnewman@newmantelecom.com>
-+ *
-+ * We would like to thank Newman Ventures, Inc. for funding this
-+ * Asterisk project.
-+ *
-+ * Newman Ventures <info@newmanventures.com>
-+ *
-+ * Portions Copyright:
-+ * Copyright (C) 2001, Linux Support Services, Inc.
-+ * Copyright (C) 2004, Digium, Inc.
-+ *
-+ * Matthew Fredrickson <creslin@linux-support.net>
-+ * Mark Spencer <markster@digium.com>
-+ *
-+ * This program is free software, distributed under the terms of
-+ * the GNU General Public License
-+ */
-+ 
-+#include "asterisk.h"
-+
-+#include <asterisk/lock.h>
-+#include <asterisk/file.h>
-+#include <asterisk/logger.h>
-+#include <asterisk/channel.h>
-+#include <asterisk/pbx.h>
-+#include <asterisk/module.h>
-+#include <asterisk/translate.h>
-+#include <asterisk/utils.h>
-+#include <asterisk/dsp.h>
-+
-+static char *app = "NVBackgroundDetect";
-+
-+static char *synopsis = "Background a file with talk and fax detect (IAX and SIP too)";
-+
-+static char *descrip = 
-+"  NVBackgroundDetect(filename[|options[|sildur[|mindur|[maxdur]]]]):\n"
-+"Plays filename, waiting for interruption from fax tones (on IAX and SIP too),\n"
-+"a digit, or non-silence. Audio is monitored in the receive direction. If\n"
-+"digits interrupt, they must be the start of a valid extension unless the\n"
-+"option is included to ignore. If fax is detected, it will jump to the\n"
-+"'fax' extension. If a period of non-silence is greater than 'mindur' ms,\n"
-+"yet less than 'maxdur' ms is followed by silence at least 'sildur' ms\n"
-+"then the app is aborted and processing jumps to the 'talk' extension.\n"
-+"If all undetected, control will continue at the next priority.\n"
-+"      options:\n"
-+"        'n':  Attempt on-hook if unanswered (default=no)\n"
-+"        'x':  DTMF digits terminate without extension (default=no)\n"
-+"        'd':  Ignore DTMF digit detection (default=no)\n"
-+"        'f':  Ignore fax detection (default=no)\n"
-+"        't':  Ignore talk detection (default=no)\n"
-+"      sildur:  Silence ms after mindur/maxdur before aborting (default=1000)\n"
-+"      mindur:  Minimum non-silence ms needed (default=100)\n"
-+"      maxdur:  Maximum non-silence ms allowed (default=0/forever)\n"
-+"Returns -1 on hangup, and 0 on successful completion with no exit conditions.\n\n"
-+"For questions or comments, please e-mail support@newmantelecom.com.\n";
-+
-+// Use the second one for recent Asterisk releases
-+#define CALLERID_FIELD cid.cid_num
-+//#define CALLERID_FIELD callerid
-+
-+static int nv_background_detect_exec(struct ast_channel *chan, void *data)
-+{
-+	int res = 0;
-+	char tmp[256] = "\0";
-+	char *p = NULL;
-+	char *filename = NULL;
-+	char *options = NULL;
-+	char *silstr = NULL;
-+	char *minstr = NULL;
-+	char *maxstr = NULL;
-+	struct ast_frame *fr = NULL;
-+	struct ast_frame *fr2 = NULL;
-+	int notsilent = 0;
-+	struct timeval start = {0, 0}, end = {0, 0};
-+	int sildur = 1000;
-+	int mindur = 100;
-+	int maxdur = -1;
-+	int skipanswer = 0;
-+	int noextneeded = 0;
-+	int ignoredtmf = 0;
-+	int ignorefax = 0;
-+	int ignoretalk = 0;
-+	int x = 0;
-+	int origrformat = 0;
-+	int features = 0;
-+	struct ast_dsp *dsp = NULL;
-+	
-+	pbx_builtin_setvar_helper(chan, "FAX_DETECTED", "");
-+	pbx_builtin_setvar_helper(chan, "FAXEXTEN", "");
-+	pbx_builtin_setvar_helper(chan, "DTMF_DETECTED", "");
-+	pbx_builtin_setvar_helper(chan, "TALK_DETECTED", "");
-+	
-+	if (!data || ast_strlen_zero((char *)data)) {
-+		ast_log(LOG_WARNING, "NVBackgroundDetect requires an argument (filename)\n");
-+		return -1;
-+	}	
-+	
-+	strncpy(tmp, (char *)data, sizeof(tmp)-1);
-+	p = tmp;
-+	
-+	filename = strsep(&p, "|");
-+	options = strsep(&p, "|");
-+	silstr = strsep(&p, "|");
-+	minstr = strsep(&p, "|");	
-+	maxstr = strsep(&p, "|");	
-+	
-+	if (options) {
-+		if (strchr(options, 'n'))
-+			skipanswer = 1;
-+		if (strchr(options, 'x'))
-+			noextneeded = 1;
-+		if (strchr(options, 'd'))
-+			ignoredtmf = 1;
-+		if (strchr(options, 'f'))
-+			ignorefax = 1;
-+		if (strchr(options, 't'))
-+			ignoretalk = 1;
-+	}
-+	
-+	if (silstr) {
-+		if ((sscanf(silstr, "%d", &x) == 1) && (x > 0))
-+			sildur = x;
-+	}
-+	
-+	if (minstr) {
-+		if ((sscanf(minstr, "%d", &x) == 1) && (x > 0))
-+			mindur = x;
-+	}
-+	
-+	if (maxstr) {
-+		if ((sscanf(maxstr, "%d", &x) == 1) && (x > 0))
-+			maxdur = x;
-+	}
-+	
-+	ast_log(LOG_DEBUG, "Preparing detect of '%s' (sildur=%dms, mindur=%dms, maxdur=%dms)\n", 
-+						tmp, sildur, mindur, maxdur);
-+						
-+	//	LOCAL_USER_ADD(u);
-+	if (chan->_state != AST_STATE_UP && !skipanswer) {
-+		/* Otherwise answer unless we're supposed to send this while on-hook */
-+		res = ast_answer(chan);
-+	}
-+	if (!res) {
-+		origrformat = chan->readformat;
-+		if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR))) 
-+			ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
-+	}
-+	if (!(dsp = ast_dsp_new())) {
-+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
-+		res = -1;
-+	}
-+
-+	if (dsp) {	
-+		if (!ignoretalk)
-+			; /* features |= DSP_FEATURE_SILENCE_SUPPRESS; */
-+		if (!ignorefax)
-+			features |= DSP_FEATURE_FAX_DETECT;
-+		//if (!ignoredtmf)
-+			features |= DSP_FEATURE_DIGIT_DETECT;
-+			
-+		ast_dsp_set_threshold(dsp, 256);
-+		ast_dsp_set_features(dsp, features | DSP_DIGITMODE_RELAXDTMF);
-+		ast_dsp_set_digitmode(dsp, DSP_DIGITMODE_DTMF);
-+	}
-+	
-+	if (!res) {
-+		ast_stopstream(chan);
-+		res = ast_streamfile(chan, tmp, chan->language);
-+		if (!res) {
-+			while(chan->stream) {
-+				res = ast_sched_wait(chan->sched);
-+				if ((res < 0) && !chan->timingfunc) {
-+					res = 0;
-+					break;
-+				}
-+				if (res < 0)
-+					res = 1000;
-+				res = ast_waitfor(chan, res);
-+				if (res < 0) {
-+					ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
-+					break;
-+				} else if (res > 0) {
-+					fr = ast_read(chan);
-+					if (!fr) {
-+						ast_log(LOG_DEBUG, "Got hangup\n");
-+						res = -1;
-+						break;
-+					}
-+					
-+					fr2 = ast_dsp_process(chan, dsp, fr);
-+					if (!fr2) {
-+						ast_log(LOG_WARNING, "Bad DSP received (what happened!!)\n");
-+						fr2 = fr;
-+					} 
-+					
-+					if (fr2->frametype == AST_FRAME_DTMF) {
-+						if (fr2->subclass == 'f' && !ignorefax) {
-+							/* Fax tone -- Handle and return NULL */
-+							ast_log(LOG_DEBUG, "Fax detected on %s\n", chan->name);
-+							if (strcmp(chan->exten, "fax")) {
-+								ast_log(LOG_NOTICE, "Redirecting %s to fax extension\n", chan->name);
-+								pbx_builtin_setvar_helper(chan, "FAX_DETECTED", "1");
-+								pbx_builtin_setvar_helper(chan,"FAXEXTEN",chan->exten);								
-+								if (ast_exists_extension(chan, chan->context, "fax", 1, chan->CALLERID_FIELD)) {
-+									/* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
-+									strncpy(chan->exten, "fax", sizeof(chan->exten)-1);
-+									chan->priority = 0;									
-+								} else
-+									ast_log(LOG_WARNING, "Fax detected, but no fax extension\n");
-+							} else
-+								ast_log(LOG_WARNING, "Already in a fax extension, not redirecting\n");
-+
-+							res = 0;
-+							ast_frfree(fr);
-+							break;
-+						} else if (!ignoredtmf) {
-+							ast_log(LOG_DEBUG, "DTMF detected on %s\n", chan->name);
-+							char t[2];
-+							t[0] = fr2->subclass;
-+							t[1] = '\0';
-+							if (noextneeded || ast_canmatch_extension(chan, chan->context, t, 1, chan->CALLERID_FIELD)) {
-+								pbx_builtin_setvar_helper(chan, "DTMF_DETECTED", "1");
-+								/* They entered a valid extension, or might be anyhow */
-+								if (noextneeded) {
-+									ast_log(LOG_NOTICE, "DTMF received (not matching to exten)\n");
-+									res = 0;
-+								} else {
-+									ast_log(LOG_NOTICE, "DTMF received (matching to exten)\n");
-+									res = fr2->subclass;
-+								}
-+								ast_frfree(fr);
-+								break;
-+							} else
-+								ast_log(LOG_DEBUG, "Valid extension requested and DTMF did not match\n");
-+						}
-+					} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR) && !ignoretalk) {
-+						int totalsilence;
-+						int ms;
-+						res = ast_dsp_silence(dsp, fr, &totalsilence);
-+						if (res && (totalsilence > sildur)) {
-+							/* We've been quiet a little while */
-+							if (notsilent) {
-+								/* We had heard some talking */
-+								gettimeofday(&end, NULL);
-+								ms = (end.tv_sec - start.tv_sec) * 1000;
-+								ms += (end.tv_usec - start.tv_usec) / 1000;
-+								ms -= sildur;
-+								if (ms < 0)
-+									ms = 0;
-+								if ((ms > mindur) && ((maxdur < 0) || (ms < maxdur))) {
-+									char ms_str[10];
-+									ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
-+									ast_log(LOG_NOTICE, "Redirecting %s to talk extension\n", chan->name);
-+
-+									/* Save detected talk time (in milliseconds) */ 
-+									sprintf(ms_str, "%d", ms);	
-+									pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
-+									
-+									if (ast_exists_extension(chan, chan->context, "talk", 1, chan->CALLERID_FIELD)) {
-+										strncpy(chan->exten, "talk", sizeof(chan->exten) - 1);
-+										chan->priority = 0;
-+									} else
-+										ast_log(LOG_WARNING, "Talk detected, but no talk extension\n");
-+									res = 0;
-+									ast_frfree(fr);
-+									break;
-+								} else
-+									ast_log(LOG_DEBUG, "Found unqualified token of %d ms\n", ms);
-+								notsilent = 0;
-+							}
-+						} else {
-+							if (!notsilent) {
-+								/* Heard some audio, mark the begining of the token */
-+								gettimeofday(&start, NULL);
-+								ast_log(LOG_DEBUG, "Start of voice token!\n");
-+								notsilent = 1;
-+							}
-+						}						
-+					}
-+					ast_frfree(fr);
-+				}
-+				ast_sched_runq(chan->sched);
-+			}
-+			ast_stopstream(chan);
-+		} else {
-+			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
-+			res = 0;
-+		}
-+	} else
-+		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
-+	
-+	if (res > -1) {
-+		if (origrformat && ast_set_read_format(chan, origrformat)) {
-+			ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n", 
-+				chan->name, ast_getformatname(origrformat));
-+		}
-+	}
-+	
-+	if (dsp)
-+		ast_dsp_free(dsp);
-+	
-+	//	LOCAL_USER_REMOVE(u);
-+	
-+	return res;
-+}
-+
-+static int unload_module(void)
-+{
-+  //	STANDARD_HANGUP_LOCALUSERS;
-+	return ast_unregister_application(app);
-+}
-+
-+static int load_module(void)
-+{
-+	return ast_register_application(app, nv_background_detect_exec, synopsis, descrip);
-+}
-+
-+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Fax playing Background Music Detection Application");
-diff -uNr apps/app_nv_faxdetect.c apps/app_nv_faxdetect.c
---- apps/app_nv_faxdetect.c	1970-01-01 01:00:00.000000000 +0100
-+++ apps/app_nv_faxdetect.c	2010-02-10 22:09:40.119390638 +0000
-@@ -0,0 +1,317 @@
-+/*
-+ * Asterisk -- A telephony toolkit for Linux.
-+ *
-+ * Fax detection application for all channel types.
-+ * 
-+ * Copyright (C) 2004-2005, Newman Telecom, Inc. and Newman Ventures, Inc.
-+ *
-+ * Justin Newman <jnewman@newmantelecom.com>
-+ *
-+ * We would like to thank Newman Ventures, Inc. for funding this
-+ * Asterisk project.
-+ *
-+ * Newman Ventures <info@newmanventures.com>
-+ *
-+ * Portions Copyright:
-+ * Copyright (C) 2001, Linux Support Services, Inc.
-+ * Copyright (C) 2004, Digium, Inc.
-+ *
-+ * Matthew Fredrickson <creslin@linux-support.net>
-+ * Mark Spencer <markster@digium.com>
-+ *
-+ * This program is free software, distributed under the terms of
-+ * the GNU General Public License
-+ */
-+ 
-+#include "asterisk.h"
-+
-+#include <asterisk/lock.h>
-+#include <asterisk/file.h>
-+#include <asterisk/logger.h>
-+#include <asterisk/channel.h>
-+#include <asterisk/pbx.h>
-+#include <asterisk/module.h>
-+#include <asterisk/translate.h>
-+#include <asterisk/dsp.h>
-+#include <asterisk/utils.h>
-+
-+static char *app = "NVFaxDetect";
-+
-+static char *synopsis = "Detects fax sounds on all channel types (IAX and SIP too)";
-+
-+static char *descrip = 
-+"  NVFaxDetect([waitdur[|options[|sildur[|mindur[|maxdur]]]]]):\n"
-+"This application listens for fax tones (on IAX and SIP channels too)\n"
-+"for waitdur seconds of time. In addition, it can be interrupted by digits,\n"
-+"or non-silence. Audio is only monitored in the receive direction. If\n"
-+"digits interrupt, they must be the start of a valid extension unless the\n"
-+"option is included to ignore. If fax is detected, it will jump to the\n"
-+"'fax' extension. If a period of non-silence greater than 'mindur' ms,\n"
-+"yet less than 'maxdur' ms is followed by silence at least 'sildur' ms\n"
-+"then the app is aborted and processing jumps to the 'talk' extension.\n"
-+"If all undetected, control will continue at the next priority.\n"
-+"      waitdur:  Maximum number of seconds to wait (default=4)\n"
-+"      options:\n"
-+"        'n':  Attempt on-hook if unanswered (default=no)\n"
-+"        'x':  DTMF digits terminate without extension (default=no)\n"
-+"        'd':  Ignore DTMF digit detection (default=no)\n"
-+"        'f':  Ignore fax detection (default=no)\n"
-+"        't':  Ignore talk detection (default=no)\n"
-+"      sildur:  Silence ms after mindur/maxdur before aborting (default=1000)\n"
-+"      mindur:  Minimum non-silence ms needed (default=100)\n"
-+"      maxdur:  Maximum non-silence ms allowed (default=0/forever)\n"
-+"Returns -1 on hangup, and 0 on successful completion with no exit conditions.\n\n"
-+"For questions or comments, please e-mail support@newmantelecom.com.\n";
-+
-+// Use the second one for recent Asterisk releases
-+#define CALLERID_FIELD cid.cid_num
-+//#define CALLERID_FIELD callerid
-+
-+static int nv_detectfax_exec(struct ast_channel *chan, void *data)
-+{
-+	int res = 0;
-+	char tmp[256] = "\0";
-+	char *p = NULL;
-+	char *waitstr = NULL;
-+	char *options = NULL;
-+	char *silstr = NULL;
-+	char *minstr = NULL;
-+	char *maxstr = NULL;
-+	struct ast_frame *fr = NULL;
-+	struct ast_frame *fr2 = NULL;
-+	int notsilent = 0;
-+	struct timeval start = {0, 0}, end = {0, 0};
-+	int waitdur = 4;
-+	int sildur = 1000;
-+	int mindur = 100;
-+	int maxdur = -1;
-+	int skipanswer = 0;
-+	int noextneeded = 0;
-+	int ignoredtmf = 0;
-+	int ignorefax = 0;
-+	int ignoretalk = 0;
-+	int x = 0;
-+	int origrformat = 0;
-+	int features = 0;
-+	time_t timeout = 0;
-+	struct ast_dsp *dsp = NULL;
-+	
-+	pbx_builtin_setvar_helper(chan, "FAX_DETECTED", "");
-+	pbx_builtin_setvar_helper(chan, "FAXEXTEN", "");
-+	pbx_builtin_setvar_helper(chan, "DTMF_DETECTED", "");
-+	pbx_builtin_setvar_helper(chan, "TALK_DETECTED", "");
-+	
-+	if (data || !ast_strlen_zero((char *)data)) {
-+		strncpy(tmp, (char *)data, sizeof(tmp)-1);
-+	}	
-+	
-+	p = tmp;
-+	
-+	waitstr = strsep(&p, "|");
-+	options = strsep(&p, "|");
-+	silstr = strsep(&p, "|");
-+	minstr = strsep(&p, "|");	
-+	maxstr = strsep(&p, "|");	
-+	
-+	if (waitstr) {
-+		if ((sscanf(waitstr, "%d", &x) == 1) && (x > 0))
-+			waitdur = x;
-+	}
-+	
-+	if (options) {
-+		if (strchr(options, 'n'))
-+			skipanswer = 1;
-+		if (strchr(options, 'x'))
-+			noextneeded = 1;
-+		if (strchr(options, 'd'))
-+			ignoredtmf = 1;
-+		if (strchr(options, 'f'))
-+			ignorefax = 1;
-+		if (strchr(options, 't'))
-+			ignoretalk = 1;
-+	}
-+	
-+	if (silstr) {
-+		if ((sscanf(silstr, "%d", &x) == 1) && (x > 0))
-+			sildur = x;
-+	}
-+	
-+	if (minstr) {
-+		if ((sscanf(minstr, "%d", &x) == 1) && (x > 0))
-+			mindur = x;
-+	}
-+	
-+	if (maxstr) {
-+		if ((sscanf(maxstr, "%d", &x) == 1) && (x > 0))
-+			maxdur = x;
-+	}
-+	
-+	ast_log(LOG_DEBUG, "Preparing detect of fax (waitdur=%dms, sildur=%dms, mindur=%dms, maxdur=%dms)\n", 
-+						waitdur, sildur, mindur, maxdur);
-+						
-+	//	LOCAL_USER_ADD(u);
-+	if (chan->_state != AST_STATE_UP && !skipanswer) {
-+		/* Otherwise answer unless we're supposed to send this while on-hook */
-+		res = ast_answer(chan);
-+	}
-+	if (!res) {
-+		origrformat = chan->readformat;
-+		if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR))) 
-+			ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
-+	}
-+	if (!(dsp = ast_dsp_new())) {
-+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
-+		res = -1;
-+	}
-+	
-+	if (dsp) {	
-+		if (!ignoretalk)
-+			; /* features |= DSP_FEATURE_SILENCE_SUPPRESS; */
-+		if (!ignorefax)
-+			features |= DSP_FEATURE_FAX_DETECT;
-+		//if (!ignoredtmf)
-+			features |= DSP_FEATURE_DIGIT_DETECT;
-+			
-+		ast_dsp_set_threshold(dsp, 256); 
-+		ast_dsp_set_features(dsp, features | DSP_DIGITMODE_RELAXDTMF);
-+		ast_dsp_set_digitmode(dsp, DSP_DIGITMODE_DTMF);
-+	}
-+
-+	if (!res) {
-+		if (waitdur > 0)
-+			timeout = time(NULL) + (time_t)waitdur;
-+
-+		while(ast_waitfor(chan, -1) > -1) {
-+			if (waitdur > 0 && time(NULL) > timeout) {
-+				res = 0;
-+				break;
-+			}
-+
-+			fr = ast_read(chan);
-+			if (!fr) {
-+				ast_log(LOG_DEBUG, "Got hangup\n");
-+				res = -1;
-+				break;
-+			}
-+
-+			fr2 = ast_dsp_process(chan, dsp, fr);
-+			if (!fr2) {
-+				ast_log(LOG_WARNING, "Bad DSP received (what happened?)\n");
-+				fr2 = fr;
-+			} 
-+
-+			if (fr2->frametype == AST_FRAME_DTMF) {
-+				if (fr2->subclass == 'f' && !ignorefax) {
-+					/* Fax tone -- Handle and return NULL */
-+					ast_log(LOG_DEBUG, "Fax detected on %s\n", chan->name);
-+					if (strcmp(chan->exten, "fax")) {
-+						ast_log(LOG_NOTICE, "Redirecting %s to fax extension\n", chan->name);
-+						pbx_builtin_setvar_helper(chan, "FAX_DETECTED", "1");
-+						pbx_builtin_setvar_helper(chan,"FAXEXTEN",chan->exten);								
-+						if (ast_exists_extension(chan, chan->context, "fax", 1, chan->CALLERID_FIELD)) {
-+							/* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
-+							strncpy(chan->exten, "fax", sizeof(chan->exten)-1);
-+							chan->priority = 0;									
-+						} else
-+							ast_log(LOG_WARNING, "Fax detected, but no fax extension\n");
-+					} else
-+						ast_log(LOG_WARNING, "Already in a fax extension, not redirecting\n");
-+
-+					res = 0;
-+					ast_frfree(fr);
-+					break;
-+				} else if (!ignoredtmf) {
-+					ast_log(LOG_DEBUG, "DTMF detected on %s\n", chan->name);
-+					char t[2];
-+					t[0] = fr2->subclass;
-+					t[1] = '\0';
-+					if (noextneeded || ast_canmatch_extension(chan, chan->context, t, 1, chan->CALLERID_FIELD)) {
-+						pbx_builtin_setvar_helper(chan, "DTMF_DETECTED", "1");
-+						/* They entered a valid extension, or might be anyhow */
-+						if (noextneeded) {
-+							ast_log(LOG_NOTICE, "DTMF received (not matching to exten)\n");
-+							res = 0;
-+						} else {
-+							ast_log(LOG_NOTICE, "DTMF received (matching to exten)\n");
-+							res = fr2->subclass;
-+						}
-+						ast_frfree(fr);
-+						break;
-+					} else
-+						ast_log(LOG_DEBUG, "Valid extension requested and DTMF did not match\n");
-+				}
-+			} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR) && !ignoretalk) {
-+				int totalsilence;
-+				int ms;
-+				res = ast_dsp_silence(dsp, fr, &totalsilence);
-+				if (res && (totalsilence > sildur)) {
-+					/* We've been quiet a little while */
-+					if (notsilent) {
-+						/* We had heard some talking */
-+						gettimeofday(&end, NULL);
-+						ms = (end.tv_sec - start.tv_sec) * 1000;
-+						ms += (end.tv_usec - start.tv_usec) / 1000;
-+						ms -= sildur;
-+						if (ms < 0)
-+							ms = 0;
-+						if ((ms > mindur) && ((maxdur < 0) || (ms < maxdur))) {
-+							char ms_str[10];
-+							ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
-+							ast_log(LOG_NOTICE, "Redirecting %s to talk extension\n", chan->name);
-+
-+							/* Save detected talk time (in milliseconds) */ 
-+							sprintf(ms_str, "%d", ms);	
-+							pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
-+
-+							if (ast_exists_extension(chan, chan->context, "talk", 1, chan->CALLERID_FIELD)) {
-+								strncpy(chan->exten, "talk", sizeof(chan->exten) - 1);
-+								chan->priority = 0;
-+							} else
-+								ast_log(LOG_WARNING, "Talk detected, but no talk extension\n");
-+							res = 0;
-+							ast_frfree(fr);
-+							break;
-+						} else
-+							ast_log(LOG_DEBUG, "Found unqualified token of %d ms\n", ms);
-+						notsilent = 0;
-+					}
-+				} else {
-+					if (!notsilent) {
-+						/* Heard some audio, mark the begining of the token */
-+						gettimeofday(&start, NULL);
-+						ast_log(LOG_DEBUG, "Start of voice token!\n");
-+						notsilent = 1;
-+					}
-+				}						
-+			}
-+			ast_frfree(fr);
-+		}
-+	} else
-+		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
-+	
-+	if (res > -1) {
-+		if (origrformat && ast_set_read_format(chan, origrformat)) {
-+			ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n", 
-+				chan->name, ast_getformatname(origrformat));
-+		}
-+	}
-+	
-+	if (dsp)
-+		ast_dsp_free(dsp);
-+	
-+	//	LOCAL_USER_REMOVE(u);
-+	
-+	return res;
-+}
-+
-+static int unload_module(void)
-+{
-+  return ast_unregister_application(app);
-+}
-+
-+static int load_module(void)
-+{
-+  return ast_register_application(app, nv_detectfax_exec, synopsis, descrip);
-+}
-+
-+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Fax Detection Application");
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-inband-indications.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-inband-indications.patch
deleted file mode 100644
index c8d409584263..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-inband-indications.patch
+++ /dev/null
@@ -1,20 +0,0 @@
---- asterisk-1.6.2.8-orig/main/channel.c	2010-06-06 10:04:33.439148782 +0200
-+++ asterisk-1.6.2.8/main/channel.c	2010-06-06 10:05:51.124146954 +0200
-@@ -2014,6 +2014,8 @@
- 			chan->generator->release(chan, chan->generatordata);
- 		chan->generatordata = NULL;
- 	}
-+
-+	ast_prod(chan);
- 	if (gen->alloc && !(chan->generatordata = gen->alloc(chan, params))) {
- 		res = -1;
- 	}
-@@ -2023,8 +2025,6 @@
- 	}
- 	ast_channel_unlock(chan);
- 
--	ast_prod(chan);
--
- 	return res;
- }
- 
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-pri-missing-keyword.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-pri-missing-keyword.patch
deleted file mode 100644
index 60cde9785d88..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.8-pri-missing-keyword.patch
+++ /dev/null
@@ -1,12 +0,0 @@
-diff -uNr asterisk-1.6.2.8.ORIG/include/asterisk/autoconfig.h.in	asterisk-1.6.2.8/include/asterisk/autoconfig.h.in
---- asterisk-1.6.2.8.ORIG/include/asterisk/autoconfig.h.in	2009-09-29 09:53:36.000000000 +0200
-+++ asterisk-1.6.2.8/include/asterisk/autoconfig.h.in	2009-09-29   2009-09-29 09:53:36.000000000 +0200
-@@ -620,6 +620,8 @@
- /* Define to 1 if you have the ISDN PRI library. */
- #undef HAVE_PRI
- 
-+#undef HAVE_PRI_PROG_W_CAUSE
-+
- /* Define to 1 if you have the ISDN PRI set_inbanddisconnect library. */
- #undef HAVE_PRI_INBANDDISCONNECT
- 
diff --git a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.9-gsm-pic.patch b/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.9-gsm-pic.patch
deleted file mode 100644
index df85382f0df0..000000000000
--- a/net-misc/asterisk/files/1.6.2/asterisk-1.6.2.9-gsm-pic.patch
+++ /dev/null
@@ -1,64 +0,0 @@
-diff -uNr asterisk-1.6.2.9.ORIG/codecs/gsm/Makefile asterisk-1.6.2.9/codecs/gsm/Makefile
---- asterisk-1.6.2.9.ORIG/codecs/gsm/Makefile	2010-03-09 21:39:20.000000000 +0100
-+++ asterisk-1.6.2.9/codecs/gsm/Makefile	2010-03-09 21:39:20.000000000 +0100
-@@ -37,30 +37,6 @@
- ######### ppro's, etc, as well as the AMD K6 and K7.  The compile will
- ######### probably require gcc. 
- 
--ifeq (,$(findstring $(OSARCH),Darwin SunOS))
--  ifeq (,$(strip $(findstring $(PROC) ,"x86_64 amd64 ultrasparc sparc64 arm armv5b armeb ppc powerpc ppc64 ia64 s390 bfin mipsel mips ")))
--    ifeq (,$(strip $(findstring $(shell uname -m) ,"ppc ppc64 alpha armv4l s390 ")))
--      OPTIMIZE+=-march=$(PROC)
--    endif
--  endif
--else
--  ifneq (,$(findstring $(OSARCH),Darwin))
--    ifeq ($(shell /usr/bin/sw_vers -productVersion | cut -c1-4),10.6)
--      # Snow Leopard reports i386, even though it's really x86_64
--      OPTIMIZE+=-mtune=native
--    endif
--  endif
--endif
--
--#The problem with sparc is the best stuff is in newer versions of gcc (post 3.0) only.
--#This works for even old (2.96) versions of gcc and provides a small boost either way.
--#A ultrasparc cpu is really v9 but the stock debian stable 3.0 gcc doesn't support it.
--#So we go lowest common available by gcc and go a step down, still a step up from
--#the default as we now have a better instruction set to work with. - Belgarath
--ifeq ($(PROC),ultrasparc)
--OPTIMIZE+=-mcpu=v8 -mtune=$(PROC) -O3 
--endif
--
- PG =
- #PG = -g -pg
- ######### Profiling flags.  If you don't know what that means, leave it blank.
-@@ -215,11 +191,9 @@
- # XXX Keep a space after each findstring argument
- # XXX should merge with GSM_OBJECTS
- ifeq ($(OSARCH),linux-gnu)
--ifeq (,$(findstring $(shell uname -m) , x86_64 amd64 ppc ppc64 alpha armv4l sparc64 parisc s390 ))
--ifeq (,$(findstring $(PROC) , arm armv5b armeb powerpc ia64 s390 bfin mipsel mips ))
--GSM_SOURCES+= $(SRC)/k6opt.s
--endif
--endif
-+  ifneq ($(K6OPT),)
-+    GSM_SOURCES+= $(SRC)/k6opt.s
-+  endif
- endif
- 
- TOAST_SOURCES = $(SRC)/toast.c 		\
-@@ -267,11 +241,9 @@
- 		$(SRC)/table.o
- 
- ifeq ($(OSARCH),linux-gnu)
--ifeq (,$(findstring $(shell uname -m) , x86_64 amd64 ppc ppc64 alpha armv4l sparc64 parisc ))
--ifeq (,$(findstring $(PROC) , arm armv5b armeb powerpc ia64 bfin mipsel mips ))
--GSM_OBJECTS+= $(SRC)/k6opt.o
--endif
--endif
-+  ifneq ($(K6OPT),)
-+    GSM_OBJECTS+= $(SRC)/k6opt.o
-+  endif
- endif
- 
- TOAST_OBJECTS =	$(SRC)/toast.o 		\