o Update to 1.4.19.2;

o move all additional functionality into separate patches and make it
opt in. It has been concern of asterisks devs that the FreeBSD pacakage
adds functionalty not present in the original version, whch could be
confusing.

17 files changed, 822 insertions, 352 deletions

diff --git a/net/asterisk/Makefile b/net/asterisk/Makefile
index f972ca541c79..c1c8e5eaa5c1 100644
--- a/net/asterisk/Makefile
+++ b/net/asterisk/Makefile
@@ -6,8 +6,7 @@
 #
 
 PORTNAME=	asterisk
-PORTVERSION=	1.4.18.1
-PORTREVISION=	1
+PORTVERSION=	1.4.19.2
 CATEGORIES=	net
 MASTER_SITES=	http://ftp.digium.com/pub/asterisk/ \
 		http://ftp.digium.com/pub/asterisk/old-releases/
@@ -50,7 +49,8 @@ OPTIONS=	OGGVORBIS	"Enable Ogg Vorbis support" on \
 		FREETDS		"Enable FreeTDS support" on \
 		JABBER		"Enable Jabber and Gtalk support" on \
 		SQLITE          "Enable SQLITE support" on \
-		CODEC_PATCH	"Apply codec negotiation patch" off
+		CODEC_PATCH	"Apply codec negotiation patch" off \
+		MISC_PATCHES	"Apply additional patches" off
 
 .include <bsd.port.pre.mk>
 
@@ -159,13 +159,23 @@ LIB_DEPENDS+=	sqlite:${PORTSDIR}/databases/sqlite2
 .endif
 
 .if defined(WITH_CODEC_PATCH)
-PATCHFILES=	asterisk-1.4.18-codec-negotiation-20080313.diff.gz
+PATCHFILES=	asterisk-1.4.19-codec-negotiation-20080408.diff.gz
 PATCH_SITES=	http://b2bua.org/chrome/site/
 EXTRA_PATCHES+=	${PATCHDIR}/codecnego-patch-Makefile
 .else
 EXTRA_PATCHES+=	${PATCHDIR}/nocodecnego-patch-Makefile
 .endif
 
+.if defined(WITH_MISC_PATCHES)
+EXTRA_PATCHES+=	${PATCHDIR}/dtmf_debug.diff
+EXTRA_PATCHES+=	${PATCHDIR}/feature_disconnect.diff
+EXTRA_PATCHES+=	${PATCHDIR}/sip_force_callid.diff
+.if defined(WITH_CODEC_PATCH)
+EXTRA_PATCHES+=	${PATCHDIR}/rtp_force_dtmf-codecnego.diff
+.else
+EXTRA_PATCHES+=	${PATCHDIR}/rtp_force_dtmf-nocodecnego.diff
+.endif
+.endif
 
 post-patch:
 	${REINPLACE_CMD} -e 's|/var/lib|${PREFIX}/share|g' ${WRKSRC}/configs/musiconhold.conf.sample
diff --git a/net/asterisk/distinfo b/net/asterisk/distinfo
index c65a3a6dbea6..64a1135cfcd3 100644
--- a/net/asterisk/distinfo
+++ b/net/asterisk/distinfo
@@ -1,6 +1,6 @@
-MD5 (asterisk-1.4.18.1.tar.gz) = a0d7ee4054a04529b745e60dd4e750c4
-SHA256 (asterisk-1.4.18.1.tar.gz) = 5ed2e8b66fc8ac96e516b0ddfea065012e39ef51c8f35ed69c6d7cd74e180d78
-SIZE (asterisk-1.4.18.1.tar.gz) = 11488923
-MD5 (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = c2bc3b82217383936961a61b239e4635
-SHA256 (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = 6a6ec9436de24e8c35d5922f7116e65e08052bc6bef6ecc12790736bcf058af5
-SIZE (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = 38966
+MD5 (asterisk-1.4.19.2.tar.gz) = a942367e20a364510881c086a3517321
+SHA256 (asterisk-1.4.19.2.tar.gz) = b3248b84ec824c3c21861607fdc0c71e4acee16edc570004ee6c9e2f712705c4
+SIZE (asterisk-1.4.19.2.tar.gz) = 11496546
+MD5 (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 12db6a4bc354d3eeeee7a900a6060b2f
+SHA256 (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 55290639455b1117541c5bf560ef8326c3e9414d266459293893b0e3d36c64f9
+SIZE (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 39004
diff --git a/net/asterisk/files/codecnego-patch-Makefile b/net/asterisk/files/codecnego-patch-Makefile
index f474578c3a35..c94aad32f623 100644
--- a/net/asterisk/files/codecnego-patch-Makefile
+++ b/net/asterisk/files/codecnego-patch-Makefile
@@ -1,6 +1,6 @@
---- Makefile.orig	Mon Dec 24 11:53:33 2007
-+++ Makefile	Mon Dec 24 11:53:33 2007
-@@ -127,7 +127,7 @@
+--- Makefile.orig	2008-04-08 14:50:57.000000000 +0300
++++ Makefile	2008-04-08 14:52:01.000000000 +0300
+@@ -129,7 +129,7 @@
    ASTMANDIR=$(mandir)
  ifneq ($(findstring BSD,$(OSARCH)),)
    ASTVARLIBDIR=$(prefix)/share/asterisk
@@ -9,7 +9,7 @@
  else
    ASTVARLIBDIR=$(localstatedir)/lib/asterisk
  endif
-@@ -222,7 +222,7 @@
+@@ -224,7 +224,7 @@
  
  ifeq ($(OSARCH),FreeBSD)
    # -V is understood by BSD Make, not by GNU make.
@@ -18,7 +18,7 @@
    ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
    AST_LIBS+=$(shell if test  $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
  endif
-@@ -402,15 +402,15 @@
+@@ -404,15 +404,15 @@
  # Should static HTTP be installed during make samples or even with its own target ala
  # webvoicemail?  There are portions here that *could* be customized but might also be
  # improved a lot.  I'll put it here for now.
@@ -39,7 +39,7 @@
  	$(MAKE) -C sounds install
  
  update: 
-@@ -431,45 +431,45 @@
+@@ -433,45 +433,45 @@
  OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
  
  bininstall: _all
@@ -115,16 +115,16 @@
  	fi
  
  $(SUBDIRS_INSTALL):
-@@ -496,7 +496,7 @@
- 		echo " WARNING WARNING WARNING" ;\
- 	fi
+@@ -505,7 +505,7 @@
+ 	@exit 1
+ endif
  
--install: datafiles bininstall $(SUBDIRS_INSTALL)
-+install: datafiles bininstall $(SUBDIRS_INSTALL) samples
+-install: badshell datafiles bininstall $(SUBDIRS_INSTALL)
++install: badshell datafiles bininstall $(SUBDIRS_INSTALL) samples
  	@if [ -x /usr/sbin/asterisk-post-install ]; then \
  		/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
  	fi
-@@ -534,31 +534,23 @@
+@@ -543,31 +543,23 @@
  upgrade: bininstall
  
  adsi:
@@ -164,7 +164,7 @@
  		( \
  		echo "[directories]" ; \
  		echo "astetcdir => $(ASTETCDIR)" ; \
-@@ -643,20 +635,23 @@
+@@ -653,20 +645,23 @@
  		echo "; Default: strict"; \
  		echo ";"; \
  		echo ";translation_algorithm = strict"; \
@@ -192,7 +192,7 @@
  	done
  	@echo " +--------- Asterisk Web Voicemail ----------+"  
  	@echo " +                                           +"
-@@ -683,10 +678,10 @@
+@@ -693,10 +688,10 @@
  
  __rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
  	rm -rf /tmp/asterisk ; \
diff --git a/net/asterisk/files/dtmf_debug.diff b/net/asterisk/files/dtmf_debug.diff
new file mode 100644
index 000000000000..81205d40dd46
--- /dev/null
+++ b/net/asterisk/files/dtmf_debug.diff
@@ -0,0 +1,225 @@
+--- channels/chan_sip.c.orig	2008-03-18 16:42:59.000000000 +0200
++++ channels/chan_sip.c	2008-03-18 17:08:34.000000000 +0200
+@@ -3768,6 +3768,7 @@
+ 		ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
+ 	else {
+ 		p->owner = newchan;
++		ast_rtp_set_chan_name(p->rtp, newchan->name);
+ 		ret = 0;
+ 	}
+ 	if (option_debug > 2)
+@@ -4032,6 +4035,7 @@
+ 	if (i->rtp) {
+ 		tmp->fds[0] = ast_rtp_fd(i->rtp);
+ 		tmp->fds[1] = ast_rtcp_fd(i->rtp);
++		ast_rtp_set_chan_id(i->rtp, i->callid);
+ 	}
+ 	if (needvideo && i->vrtp) {
+ 		tmp->fds[2] = ast_rtp_fd(i->vrtp);
+@@ -4059,6 +4063,8 @@
+ 	if (!ast_strlen_zero(i->language))
+ 		ast_string_field_set(tmp, language, i->language);
+ 	i->owner = tmp;
++	ast_rtp_set_chan_name(i->rtp, tmp->name);
++
+ 	ast_module_ref(ast_module_info->self);
+ 	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
+ 	/*Since it is valid to have extensions in the dialplan that have unescaped characters in them
+@@ -4479,8 +4485,10 @@
+ 	build_via(p);
+ 	if (!callid)
+ 		build_callid_pvt(p);
+-	else
++	else {
+ 		ast_string_field_set(p, callid, callid);
++		ast_rtp_set_chan_id(p->rtp, p->callid);
++	}
+ 	/* Assign default music on hold class */
+ 	ast_string_field_set(p, mohinterpret, default_mohinterpret);
+ 	ast_string_field_set(p, mohsuggest, default_mohsuggest);
+--- include/asterisk/rtp.h.orig	2008-03-18 13:35:42.000000000 +0200
++++ include/asterisk/rtp.h	2008-03-18 13:35:58.000000000 +0200
+@@ -251,6 +251,9 @@
+ 
+ int ast_rtp_codec_getformat(int pt);
+ 
++void ast_rtp_set_chan_name(struct ast_rtp *, const char *);
++void ast_rtp_set_chan_id(struct ast_rtp *, const char *);
++
+ /*! \brief Set rtp timeout */
+ void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+ /*! \brief Set rtp hold timeout */
+--- main/rtp.c.orig	2008-04-08 14:53:18.000000000 +0300
++++ main/rtp.c	2008-04-08 14:54:14.000000000 +0300
+@@ -81,6 +81,7 @@
+ static int rtpstart;			/*!< First port for RTP sessions (set in rtp.conf) */
+ static int rtpend;			/*!< Last port for RTP sessions (set in rtp.conf) */
+ static int rtpdebug;			/*!< Are we debugging? */
++static int rtpdebugdtmf;		/*!< Are we debugging DTMFs? */
+ static int rtcpdebug;			/*!< Are we debugging RTCP? */
+ static int rtcpstats;			/*!< Are we debugging RTCP? */
+ static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
+@@ -168,6 +169,8 @@
+ 	struct ast_codec_pref pref;
+ 	struct ast_rtp *bridged;        /*!< Who we are Packet bridged to */
+ 	int set_marker_bit:1;           /*!< Whether to set the marker bit or not */
++	char chan_name[100];
++	char chan_id[100];
+ };
+ 
+ /* Forward declarations */
+@@ -669,8 +672,8 @@
+ 	struct ast_frame *f = NULL;
+ 	event = ntohl(*((unsigned int *)(data)));
+ 	event &= 0x001F;
+-	if (option_debug > 2 || rtpdebug)
+-		ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
++	if (option_debug > 2 || rtpdebug || rtpdebugdtmf)
++		ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF packet: %08x (len = %d)\n", rtp->chan_name, rtp->chan_id, event, len);
+ 	if (event < 10) {
+ 		resp = '0' + event;
+ 	} else if (event < 11) {
+@@ -684,12 +687,24 @@
+ 	}
+ 	if (rtp->resp && (rtp->resp != resp)) {
+ 		f = send_dtmf(rtp, AST_FRAME_DTMF_END);
++		ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF event: %c\n", rtp->chan_name, rtp->chan_id, rtp->resp);
+ 	}
+ 	rtp->resp = resp;
+ 	rtp->dtmfcount = dtmftimeout;
+ 	return f;
+ }
+ 
++void ast_rtp_set_chan_id(struct ast_rtp *rtp, const char *chan_id) {
++	if (rtp == NULL || chan_id == NULL)
++		return;
++	snprintf(rtp->chan_id, sizeof(rtp->chan_id), "%s", chan_id);
++}
++
++void ast_rtp_set_chan_name(struct ast_rtp *rtp, const char *chan_name) {
++	if (rtp == NULL || chan_name == NULL)
++		return;
++	snprintf(rtp->chan_name, sizeof(rtp->chan_name), "%s", chan_name);
++}
+ /*! 
+  * \brief Process RTP DTMF and events according to RFC 2833.
+  * 
+@@ -1051,6 +1066,10 @@
+ 	struct rtpPayloadType rtpPT;
+ 	int reconstruct = ntohl(rtpheader[0]);
+ 
++	/* If we are listening for DTMF - then feed all packets into the core  to keep the RTP stream consistent when relaying DTMFs */
++	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF))
++		return -1;
++
+ 	/* Get fields from packet */
+ 	payload = (reconstruct & 0x7f0000) >> 16;
+ 	mark = (((reconstruct & 0x800000) >> 23) != 0);
+@@ -1062,10 +1081,6 @@
+ 	if (!bridged->current_RTP_PT[payload].code)
+ 		return -1;
+ 
+-	/* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
+-	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
+-		return -1;
+-
+ 	/* Otherwise adjust bridged payload to match */
+ 	bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
+ 
+@@ -1254,11 +1269,12 @@
+ 		/* This is special in-band data that's not one of our codecs */
+ 		if (rtpPT.code == AST_RTP_DTMF) {
+ 			/* It's special -- rfc2833 process it */
+-			if (rtp_debug_test_addr(&sin)) {
++			if (rtp_debug_test_addr(&sin) || rtpdebugdtmf) {
+ 				unsigned char *data;
+ 				unsigned int event;
+ 				unsigned int event_end;
+ 				unsigned int duration;
++
+ 				data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
+ 				event = ntohl(*((unsigned int *)(data)));
+ 				event >>= 24;
+@@ -1267,9 +1283,12 @@
+ 				event_end >>= 24;
+ 				duration = ntohl(*((unsigned int *)(data)));
+ 				duration &= 0xFFFF;
+-				ast_verbose("Got  RTP RFC2833 from   %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
++
++				ast_verbose("Channel: %s %s Got RTP RFC2833 from %s:%u to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d)\n", rtp->chan_name, rtp->chan_id, ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->us.sin_addr), ntohs(rtp->us.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
+ 			}
+ 			f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
++			if (rtpdebugdtmf && f)
++				ast_verbose("Channel: %s %s Got RFC2833 DTMF event %c of type %s\n", rtp->chan_name, rtp->chan_id, f->subclass, (f->frametype == AST_FRAME_DTMF_BEGIN ? "DTMF BEGIN" : (f->frametype == AST_FRAME_DTMF_END ? "DTMF_END" : "UNKNOWN")));
+ 		} else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
+ 			/* It's really special -- process it the Cisco way */
+ 			if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
+@@ -2198,8 +2217,9 @@
+ 			ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
+ 				ast_inet_ntoa(rtp->them.sin_addr),
+ 				ntohs(rtp->them.sin_port), strerror(errno));
+-		if (rtp_debug_test_addr(&rtp->them))
+-			ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++		if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++			ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++				    rtp->chan_name, rtp->chan_id,
+ 				    ast_inet_ntoa(rtp->them.sin_addr),
+ 				    ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ 		/* Increment sequence number */
+@@ -2242,8 +2262,9 @@
+ 		ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ 			ast_inet_ntoa(rtp->them.sin_addr),
+ 			ntohs(rtp->them.sin_port), strerror(errno));
+-	if (rtp_debug_test_addr(&rtp->them))
+-		ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++	if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++		ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++			    rtp->chan_name, rtp->chan_id,
+ 			    ast_inet_ntoa(rtp->them.sin_addr),
+ 			    ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ 
+@@ -3481,6 +3502,16 @@
+ 	return RESULT_SUCCESS;
+ }
+ 
++static int rtp_do_debug_dtmf(int fd, int argc, char *argv[])
++{
++	if (argc != 3)
++		return RESULT_SHOWUSAGE;
++
++	rtpdebugdtmf = 1;
++	ast_cli(fd, "RTP DTMF debugging enabled\n");
++	return RESULT_SUCCESS;
++}
++
+ static int rtp_do_debug(int fd, int argc, char *argv[])
+ {
+ 	if (argc != 2) {
+@@ -3541,6 +3572,7 @@
+ 	if (argc != 3)
+ 		return RESULT_SHOWUSAGE;
+ 	rtpdebug = 0;
++	rtpdebugdtmf = 0;
+ 	ast_cli(fd,"RTP Debugging Disabled\n");
+ 	return RESULT_SUCCESS;
+ }
+@@ -3601,7 +3633,7 @@
+ }
+ 
+ static char debug_usage[] =
+-  "Usage: rtp debug [ip host[:port]]\n"
++  "Usage: rtp debug [ip host[:port] | dtmf]\n"
+   "       Enable dumping of all RTP packets to and from host.\n";
+ 
+ static char no_debug_usage[] =
+@@ -3676,6 +3708,10 @@
+ 	rtp_do_debug, "Enable RTP debugging",
+ 	debug_usage },
+ 
++	{ { "rtp", "debug", "dtmf", NULL },
++	rtp_do_debug_dtmf, "Enable RTP debugging on DTMFs",
++	debug_usage },
++
+ 	{ { "rtp", "debug", "off", NULL },
+ 	rtp_no_debug, "Disable RTP debugging",
+ 	no_debug_usage, NULL, &cli_rtp_no_debug_deprecated },
diff --git a/net/asterisk/files/feature_disconnect.diff b/net/asterisk/files/feature_disconnect.diff
new file mode 100644
index 000000000000..5be713b0da13
--- /dev/null
+++ b/net/asterisk/files/feature_disconnect.diff
@@ -0,0 +1,292 @@
+--- apps/app_dial.c.orig	Wed Feb  6 18:45:30 2008
++++ apps/app_dial.c	Wed Feb  6 18:51:59 2008
+@@ -297,6 +297,8 @@ AST_APP_OPTIONS(dial_exec_options, {
+ 	OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
+ 	OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
+ 
++static int detect_disconnect(struct ast_channel *chan, char code);
++
+ /* We define a custom "local user" structure because we
+    use it not only for keeping track of what is in use but
+    also for keeping track of who we're dialing. */
+@@ -402,6 +404,7 @@ static struct ast_channel *wait_for_answ
+ 	struct ast_channel *peer = NULL;
+ 	/* single is set if only one destination is enabled */
+ 	int single = outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
++
+ 	
+ 	if (single) {
+ 		/* Turn off hold music, etc */
+@@ -716,9 +719,9 @@ static struct ast_channel *wait_for_answ
+ 				}
+ 
+ 				if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) && 
+-						  (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
++						  detect_disconnect(in, f->subclass)) { /* hmm it it not guaranteed to be '*' anymore. */
+ 					if (option_verbose > 2)
+-						ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
++						ast_verbose(VERBOSE_PREFIX_3 "User requested call disconnect.\n");
+ 					*to=0;
+ 					ast_cdr_noanswer(in->cdr);
+ 					strcpy(status, "CANCEL");
+@@ -776,6 +779,56 @@ static struct ast_channel *wait_for_answ
+ 	
+ 	return peer;
+ }
++
++static char featurecode[FEATURE_MAX_LEN + 1] = "";
++
++static int detect_disconnect(struct ast_channel *chan, char code)
++{
++	struct feature_interpret_result result;
++	int x;
++	struct ast_flags features;
++	int res = FEATURE_RETURN_PASSDIGITS;
++	struct ast_call_feature *feature;
++	char *cptr;
++	int len;
++
++	len = strlen(featurecode);
++	if (len >= FEATURE_MAX_LEN) {
++		featurecode[0] = '\0';
++	}
++	cptr = &featurecode[strlen(featurecode)];
++	cptr[0] = code;
++	cptr[1] = '\0';
++
++	memset(&features, 0, sizeof(struct ast_flags));
++	ast_set_flag(&features, AST_FEATURE_DISCONNECT);
++
++	ast_features_lock();
++
++	res = ast_feature_detect(chan, &features, featurecode, &result);
++
++	if (res != FEATURE_RETURN_STOREDIGITS)
++		featurecode[0] = '\0';
++
++	if (result.builtin_feature && result.builtin_feature->feature_mask & AST_FEATURE_DISCONNECT) {
++		ast_features_unlock();
++		return 1;
++	}
++		
++	for (x = 0; x < result.num_dyn_features; ++x) {
++		feature = result.dynamic_features[x];
++		if (feature->feature_mask & AST_FEATURE_DISCONNECT) {
++			ast_features_unlock();
++			return 1;
++		}
++	}
++
++	ast_features_unlock();
++
++	return 0;
++}
++
++
+ 
+ static void replace_macro_delimiter(char *s)
+ {
+--- include/asterisk/features.h.orig	Thu Aug 23 23:16:41 2007
++++ include/asterisk/features.h	Tue Dec 11 17:13:52 2007
+@@ -31,6 +31,20 @@
+ #define FEATURE_EXTEN_LEN	32
+ #define FEATURE_MOH_LEN		80  /* same as MAX_MUSICCLASS from channel.h */
+ 
++#define FEATURE_RETURN_HANGUP		-1
++#define FEATURE_RETURN_SUCCESSBREAK	 0
++#define FEATURE_RETURN_PBX_KEEPALIVE	AST_PBX_KEEPALIVE
++#define FEATURE_RETURN_NO_HANGUP_PEER	AST_PBX_NO_HANGUP_PEER
++#define FEATURE_RETURN_PASSDIGITS	 21
++#define FEATURE_RETURN_STOREDIGITS	 22
++#define FEATURE_RETURN_SUCCESS	 	 23
++#define FEATURE_RETURN_KEEPTRYING    24
++
++#define FEATURE_SENSE_CHAN	(1 << 0)
++#define FEATURE_SENSE_PEER	(1 << 1)
++
++typedef int (*feature_operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
++
+ /*! \brief main call feature structure */
+ struct ast_call_feature {
+ 	int feature_mask;
+@@ -38,7 +52,7 @@
+ 	char sname[FEATURE_SNAME_LEN];
+ 	char exten[FEATURE_MAX_LEN];
+ 	char default_exten[FEATURE_MAX_LEN];
+-	int (*operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
++	feature_operation operation;
+ 	unsigned int flags;
+ 	char app[FEATURE_APP_LEN];		
+ 	char app_args[FEATURE_APP_ARGS_LEN];
+@@ -47,6 +61,12 @@
+ };
+ 
+ 
++struct feature_interpret_result {
++    struct ast_call_feature *builtin_feature;
++    struct ast_call_feature *dynamic_features[20];
++    int num_dyn_features;
++};
++
+ 
+ /*! \brief Park a call and read back parked location 
+  *  \param chan the channel to actually be parked
+@@ -93,5 +113,11 @@
+ /*! \brief unregister feature from feature_set
+     \param feature the ast_call_feature object which was registered before*/
+ void ast_unregister_feature(struct ast_call_feature *feature);
++
++int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result);
++
++void ast_features_lock(void);
++void ast_features_unlock(void);
++
+ 
+ #endif /* _AST_FEATURES_H */
+--- res/res_features.c.orig	2008-04-08 14:55:25.000000000 +0300
++++ res/res_features.c	2008-04-08 14:59:59.000000000 +0300
+@@ -485,18 +485,6 @@
+ }
+ 
+ 
+-#define FEATURE_RETURN_HANGUP		-1
+-#define FEATURE_RETURN_SUCCESSBREAK	 0
+-#define FEATURE_RETURN_PBX_KEEPALIVE	AST_PBX_KEEPALIVE
+-#define FEATURE_RETURN_NO_HANGUP_PEER	AST_PBX_NO_HANGUP_PEER
+-#define FEATURE_RETURN_PASSDIGITS	 21
+-#define FEATURE_RETURN_STOREDIGITS	 22
+-#define FEATURE_RETURN_SUCCESS	 	 23
+-#define FEATURE_RETURN_KEEPTRYING    24
+-
+-#define FEATURE_SENSE_CHAN	(1 << 0)
+-#define FEATURE_SENSE_PEER	(1 << 1)
+-
+ /*! \brief
+  * set caller and callee according to the direction
+  */
+@@ -1061,32 +1049,35 @@
+ 	return res;
+ }
+ 
+-static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
++void ast_features_lock(void)
++{
++	ast_rwlock_rdlock(&features_lock);
++	AST_LIST_LOCK(&feature_list);
++}
++
++void ast_features_unlock(void)
++{
++	AST_LIST_UNLOCK(&feature_list);
++	ast_rwlock_unlock(&features_lock);
++}
++
++int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result)
+ {
+ 	int x;
+-	struct ast_flags features;
+ 	int res = FEATURE_RETURN_PASSDIGITS;
+ 	struct ast_call_feature *feature;
+-	const char *dynamic_features;
++	const char *dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
+ 	char *tmp, *tok;
+ 
+-	if (sense == FEATURE_SENSE_CHAN) {
+-		ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);	
+-		dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
+-	} else {
+-		ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);	
+-		dynamic_features = pbx_builtin_getvar_helper(peer, "DYNAMIC_FEATURES");
+-	}
+-	if (option_debug > 2)
+-		ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, code=%s, sense=%d, features=%d dynamic=%s\n", chan->name, peer->name, code, sense, features.flags, dynamic_features);
++	result->builtin_feature = NULL;
++	result->num_dyn_features = 0;
+ 
+-	ast_rwlock_rdlock(&features_lock);
+ 	for (x = 0; x < FEATURES_COUNT; x++) {
+-		if ((ast_test_flag(&features, builtin_features[x].feature_mask)) &&
++		if ((ast_test_flag(features, builtin_features[x].feature_mask)) &&
+ 		    !ast_strlen_zero(builtin_features[x].exten)) {
+ 			/* Feature is up for consideration */
+ 			if (!strcmp(builtin_features[x].exten, code)) {
+-				res = builtin_features[x].operation(chan, peer, config, code, sense, NULL);
++				result->builtin_feature = &builtin_features[x];
+ 				break;
+ 			} else if (!strncmp(builtin_features[x].exten, code, strlen(code))) {
+ 				if (res == FEATURE_RETURN_PASSDIGITS)
+@@ -1094,7 +1085,6 @@
+ 			}
+ 		}
+ 	}
+-	ast_rwlock_unlock(&features_lock);
+ 
+ 	if (ast_strlen_zero(dynamic_features))
+ 		return res;
+@@ -1102,9 +1092,7 @@
+ 	tmp = ast_strdupa(dynamic_features);
+ 
+ 	while ((tok = strsep(&tmp, "#"))) {
+-		AST_LIST_LOCK(&feature_list);	
+ 		if (!(feature = find_dynamic_feature(tok))) {
+-			AST_LIST_UNLOCK(&feature_list);
+ 			continue;
+ 		}
+ 			
+@@ -1112,21 +1100,52 @@
+ 		if (!strcmp(feature->exten, code)) {
+ 			if (option_verbose > 2)
+ 				ast_verbose(VERBOSE_PREFIX_3 " Feature Found: %s exten: %s\n",feature->sname, tok);
+-			res = feature->operation(chan, peer, config, code, sense, feature);
+-			if (res != FEATURE_RETURN_KEEPTRYING) {
+-				AST_LIST_UNLOCK(&feature_list);
++			result->dynamic_features[result->num_dyn_features++] = feature;
++			if (result->num_dyn_features >= (sizeof(result->dynamic_features) / sizeof(result->dynamic_features[0]))) {
+ 				break;
+ 			}
+ 			res = FEATURE_RETURN_PASSDIGITS;
+ 		} else if (!strncmp(feature->exten, code, strlen(code)))
+ 			res = FEATURE_RETURN_STOREDIGITS;
+ 
+-		AST_LIST_UNLOCK(&feature_list);
+ 	}
+ 	
+ 	return res;
+ }
+ 
++static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
++{
++	struct feature_interpret_result result;
++	int x;
++	struct ast_flags features;
++	int res = FEATURE_RETURN_PASSDIGITS;
++	struct ast_call_feature *feature;
++
++	if (sense == FEATURE_SENSE_CHAN)
++		ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);	
++	else
++		ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);	
++	if (option_debug > 2)
++		ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d\n", chan->name, peer->name, sense, features.flags);
++
++	ast_features_lock();
++	res = ast_feature_detect(chan, &features, code, &result);
++
++	if (result.builtin_feature)
++		res = result.builtin_feature->operation(chan, peer, config, code, sense, NULL);
++
++	for (x = 0; x < result.num_dyn_features; ++x) {
++		feature = result.dynamic_features[x];
++		res = feature->operation(chan, peer, config, code, sense, feature);
++		if (res != FEATURE_RETURN_KEEPTRYING)
++			break;
++		res = FEATURE_RETURN_PASSDIGITS;
++	}
++
++	ast_features_unlock();
++	return res;
++}
++
+ static void set_config_flags(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
+ {
+ 	int x;
diff --git a/net/asterisk/files/nocodecnego-patch-Makefile b/net/asterisk/files/nocodecnego-patch-Makefile
index f73c3d83bcde..1d9675f1d664 100644
--- a/net/asterisk/files/nocodecnego-patch-Makefile
+++ b/net/asterisk/files/nocodecnego-patch-Makefile
@@ -1,6 +1,6 @@
---- Makefile.orig	Mon Dec 17 08:44:51 2007
-+++ Makefile	Mon Dec 24 11:54:52 2007
-@@ -127,7 +127,7 @@
+--- Makefile.orig	2008-03-25 16:37:35.000000000 +0200
++++ Makefile	2008-04-08 15:02:12.000000000 +0300
+@@ -129,7 +129,7 @@
    ASTMANDIR=$(mandir)
  ifneq ($(findstring BSD,$(OSARCH)),)
    ASTVARLIBDIR=$(prefix)/share/asterisk
@@ -9,7 +9,7 @@
  else
    ASTVARLIBDIR=$(localstatedir)/lib/asterisk
  endif
-@@ -222,7 +222,7 @@
+@@ -224,7 +224,7 @@
  
  ifeq ($(OSARCH),FreeBSD)
    # -V is understood by BSD Make, not by GNU make.
@@ -18,7 +18,7 @@
    ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
    AST_LIBS+=$(shell if test  $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
  endif
-@@ -402,15 +402,15 @@
+@@ -404,15 +404,15 @@
  # Should static HTTP be installed during make samples or even with its own target ala
  # webvoicemail?  There are portions here that *could* be customized but might also be
  # improved a lot.  I'll put it here for now.
@@ -39,7 +39,7 @@
  	$(MAKE) -C sounds install
  
  update: 
-@@ -431,45 +431,45 @@
+@@ -433,45 +433,45 @@
  OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
  
  bininstall: _all
@@ -115,16 +115,16 @@
  	fi
  
  $(SUBDIRS_INSTALL):
-@@ -496,7 +496,7 @@
- 		echo " WARNING WARNING WARNING" ;\
- 	fi
+@@ -505,7 +505,7 @@
+ 	@exit 1
+ endif
  
--install: datafiles bininstall $(SUBDIRS_INSTALL)
-+install: datafiles bininstall $(SUBDIRS_INSTALL) samples
+-install: badshell datafiles bininstall $(SUBDIRS_INSTALL)
++install: badshell datafiles bininstall $(SUBDIRS_INSTALL) samples
  	@if [ -x /usr/sbin/asterisk-post-install ]; then \
  		/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
  	fi
-@@ -534,31 +534,23 @@
+@@ -543,31 +543,23 @@
  upgrade: bininstall
  
  adsi:
@@ -164,7 +164,7 @@
  		( \
  		echo "[directories]" ; \
  		echo "astetcdir => $(ASTETCDIR)" ; \
-@@ -602,20 +594,23 @@
+@@ -612,20 +604,23 @@
  		echo ";astctlowner = root" ; \
  		echo ";astctlgroup = apache" ; \
  		echo ";astctl = asterisk.ctl" ; \
@@ -192,7 +192,7 @@
  	done
  	@echo " +--------- Asterisk Web Voicemail ----------+"  
  	@echo " +                                           +"
-@@ -642,10 +637,10 @@
+@@ -652,10 +647,10 @@
  
  __rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
  	rm -rf /tmp/asterisk ; \
diff --git a/net/asterisk/files/patch-apps::app_dial.c b/net/asterisk/files/patch-apps::app_dial.c
deleted file mode 100644
index 6abe48a17d9f..000000000000
--- a/net/asterisk/files/patch-apps::app_dial.c
+++ /dev/null
@@ -1,88 +0,0 @@
---- apps/app_dial.c.orig	Wed Feb  6 18:45:30 2008
-+++ apps/app_dial.c	Wed Feb  6 18:51:59 2008
-@@ -297,6 +297,8 @@ AST_APP_OPTIONS(dial_exec_options, {
- 	OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
- 	OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
- 
-+static int detect_disconnect(struct ast_channel *chan, char code);
-+
- /* We define a custom "local user" structure because we
-    use it not only for keeping track of what is in use but
-    also for keeping track of who we're dialing. */
-@@ -402,6 +404,7 @@ static struct ast_channel *wait_for_answ
- 	struct ast_channel *peer = NULL;
- 	/* single is set if only one destination is enabled */
- 	int single = outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
-+
- 	
- 	if (single) {
- 		/* Turn off hold music, etc */
-@@ -716,9 +719,9 @@ static struct ast_channel *wait_for_answ
- 				}
- 
- 				if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) && 
--						  (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
-+						  detect_disconnect(in, f->subclass)) { /* hmm it it not guaranteed to be '*' anymore. */
- 					if (option_verbose > 2)
--						ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
-+						ast_verbose(VERBOSE_PREFIX_3 "User requested call disconnect.\n");
- 					*to=0;
- 					ast_cdr_noanswer(in->cdr);
- 					strcpy(status, "CANCEL");
-@@ -776,6 +779,56 @@ static struct ast_channel *wait_for_answ
- 	
- 	return peer;
- }
-+
-+static char featurecode[FEATURE_MAX_LEN + 1] = "";
-+
-+static int detect_disconnect(struct ast_channel *chan, char code)
-+{
-+	struct feature_interpret_result result;
-+	int x;
-+	struct ast_flags features;
-+	int res = FEATURE_RETURN_PASSDIGITS;
-+	struct ast_call_feature *feature;
-+	char *cptr;
-+	int len;
-+
-+	len = strlen(featurecode);
-+	if (len >= FEATURE_MAX_LEN) {
-+		featurecode[0] = '\0';
-+	}
-+	cptr = &featurecode[strlen(featurecode)];
-+	cptr[0] = code;
-+	cptr[1] = '\0';
-+
-+	memset(&features, 0, sizeof(struct ast_flags));
-+	ast_set_flag(&features, AST_FEATURE_DISCONNECT);
-+
-+	ast_features_lock();
-+
-+	res = ast_feature_detect(chan, &features, featurecode, &result);
-+
-+	if (res != FEATURE_RETURN_STOREDIGITS)
-+		featurecode[0] = '\0';
-+
-+	if (result.builtin_feature && result.builtin_feature->feature_mask & AST_FEATURE_DISCONNECT) {
-+		ast_features_unlock();
-+		return 1;
-+	}
-+		
-+	for (x = 0; x < result.num_dyn_features; ++x) {
-+		feature = result.dynamic_features[x];
-+		if (feature->feature_mask & AST_FEATURE_DISCONNECT) {
-+			ast_features_unlock();
-+			return 1;
-+		}
-+	}
-+
-+	ast_features_unlock();
-+
-+	return 0;
-+}
-+
-+
- 
- static void replace_macro_delimiter(char *s)
- {
diff --git a/net/asterisk/files/patch-channels::chan_sip.c b/net/asterisk/files/patch-channels::chan_sip.c
index 916cc7ffa0ac..e54df0eb1bf1 100644
--- a/net/asterisk/files/patch-channels::chan_sip.c
+++ b/net/asterisk/files/patch-channels::chan_sip.c
@@ -1,6 +1,6 @@
---- channels/chan_sip.c.orig	Mon Dec 24 11:59:46 2007
-+++ channels/chan_sip.c	Mon Dec 24 11:58:47 2007
-@@ -493,7 +493,7 @@
+--- channels/chan_sip.c.orig	2008-03-18 16:42:59.000000000 +0200
++++ channels/chan_sip.c	2008-03-18 17:08:34.000000000 +0200
+@@ -495,7 +495,7 @@
  #define DEFAULT_MOHINTERPRET    "default"
  #define DEFAULT_MOHSUGGEST      ""
  #define DEFAULT_VMEXTEN 	"asterisk"
@@ -9,16 +9,7 @@
  #define DEFAULT_NOTIFYMIME 	"application/simple-message-summary"
  #define DEFAULT_MWITIME 	10
  #define DEFAULT_ALLOWGUEST	TRUE
-@@ -3985,6 +3985,8 @@
- 	ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
- 	fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
- 
-+	pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
-+
- 	/* If we have a prefcodec setting, we have an inbound channel that set a 
- 	   preferred format for this call. Otherwise, we check the jointcapability
- 	   We also check for vrtp. If it's not there, we are not allowed do any video anyway.
-@@ -15845,6 +15847,9 @@
+@@ -15873,6 +15881,9 @@
  	char *ext, *host;
  	char tmp[256];
  	char *dest = data;
@@ -26,9 +17,9 @@
 +	char *md5secret = NULL;
 +	char *authname = NULL;
  
- 	if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
- 		ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data);
-@@ -15866,6 +15871,17 @@
+ 	oldformat = format;
+ 	if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
+@@ -15903,6 +15914,17 @@
  	if (host) {
  		*host++ = '\0';
  		ext = tmp;
@@ -46,7 +37,7 @@
  	} else {
  		ext = strchr(tmp, '/');
  		if (ext) 
-@@ -15898,6 +15914,14 @@
+@@ -15933,6 +15955,14 @@
  		ast_string_field_set(p, username, ext);
  		ast_string_field_free(p, fullcontact);
  	}
diff --git a/net/asterisk/files/patch-channels::h323::ast_h323.cxx b/net/asterisk/files/patch-channels::h323::ast_h323.cxx
new file mode 100644
index 000000000000..bfc819452128
--- /dev/null
+++ b/net/asterisk/files/patch-channels::h323::ast_h323.cxx
@@ -0,0 +1,29 @@
+--- channels/h323/ast_h323.cxx.orig	2008-04-08 15:08:37.000000000 +0300
++++ channels/h323/ast_h323.cxx	2008-04-08 15:08:51.000000000 +0300
+@@ -81,8 +81,6 @@
+ /** PWLib entry point */
+ static MyProcess *localProcess = NULL;
+ 
+-static int _timerChangePipe[2];
+-
+ static unsigned traceOptions = PTrace::Timestamp | PTrace::Thread | PTrace::FileAndLine;
+ 
+ class PAsteriskLog : public PObject, public iostream {
+@@ -191,8 +189,6 @@
+ 
+ MyProcess::~MyProcess()
+ {
+-	_timerChangePipe[0] = timerChangePipe[0];
+-	_timerChangePipe[1] = timerChangePipe[1];
+ }
+ 
+ void MyProcess::Main()
+@@ -2460,8 +2456,6 @@
+ 	if (localProcess) {
+ 		delete localProcess;
+ 		localProcess = NULL;
+-		close(_timerChangePipe[0]);
+-		close(_timerChangePipe[1]);
+ 	}
+ 	if (logstream) {
+ 		PTrace::SetLevel(0);
diff --git a/net/asterisk/files/patch-include::asterisk::features.h b/net/asterisk/files/patch-include::asterisk::features.h
deleted file mode 100644
index fa5c747eb316..000000000000
--- a/net/asterisk/files/patch-include::asterisk::features.h
+++ /dev/null
@@ -1,57 +0,0 @@
---- include/asterisk/features.h.orig	Thu Aug 23 23:16:41 2007
-+++ include/asterisk/features.h	Tue Dec 11 17:13:52 2007
-@@ -31,6 +31,20 @@
- #define FEATURE_EXTEN_LEN	32
- #define FEATURE_MOH_LEN		80  /* same as MAX_MUSICCLASS from channel.h */
- 
-+#define FEATURE_RETURN_HANGUP		-1
-+#define FEATURE_RETURN_SUCCESSBREAK	 0
-+#define FEATURE_RETURN_PBX_KEEPALIVE	AST_PBX_KEEPALIVE
-+#define FEATURE_RETURN_NO_HANGUP_PEER	AST_PBX_NO_HANGUP_PEER
-+#define FEATURE_RETURN_PASSDIGITS	 21
-+#define FEATURE_RETURN_STOREDIGITS	 22
-+#define FEATURE_RETURN_SUCCESS	 	 23
-+#define FEATURE_RETURN_KEEPTRYING    24
-+
-+#define FEATURE_SENSE_CHAN	(1 << 0)
-+#define FEATURE_SENSE_PEER	(1 << 1)
-+
-+typedef int (*feature_operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
-+
- /*! \brief main call feature structure */
- struct ast_call_feature {
- 	int feature_mask;
-@@ -38,7 +52,7 @@
- 	char sname[FEATURE_SNAME_LEN];
- 	char exten[FEATURE_MAX_LEN];
- 	char default_exten[FEATURE_MAX_LEN];
--	int (*operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
-+	feature_operation operation;
- 	unsigned int flags;
- 	char app[FEATURE_APP_LEN];		
- 	char app_args[FEATURE_APP_ARGS_LEN];
-@@ -47,6 +61,12 @@
- };
- 
- 
-+struct feature_interpret_result {
-+    struct ast_call_feature *builtin_feature;
-+    struct ast_call_feature *dynamic_features[20];
-+    int num_dyn_features;
-+};
-+
- 
- /*! \brief Park a call and read back parked location 
-  *  \param chan the channel to actually be parked
-@@ -93,5 +113,11 @@
- /*! \brief unregister feature from feature_set
-     \param feature the ast_call_feature object which was registered before*/
- void ast_unregister_feature(struct ast_call_feature *feature);
-+
-+int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result);
-+
-+void ast_features_lock(void);
-+void ast_features_unlock(void);
-+
- 
- #endif /* _AST_FEATURES_H */
diff --git a/net/asterisk/files/patch-main::Makefile b/net/asterisk/files/patch-main::Makefile
index 3d1365ea46fa..56624065f61f 100644
--- a/net/asterisk/files/patch-main::Makefile
+++ b/net/asterisk/files/patch-main::Makefile
@@ -1,8 +1,5 @@
-
-$FreeBSD$
-
---- main/Makefile.orig
-+++ main/Makefile
+--- main/Makefile.orig	2008-01-29 19:43:41.000000000 +0200
++++ main/Makefile	2008-03-13 13:57:48.000000000 +0200
 @@ -102,8 +102,8 @@
  	cd editline && test -f config.h || CFLAGS="$(PTHREAD_CFLAGS) $(subst $(ASTTOPDIR),../../,$(ASTCFLAGS:-Werror=))" LDFLAGS="$(ASTLDFLAGS)" ./configure --build=$(BUILD_PLATFORM) --host=$(HOST_PLATFORM) --with-ncurses=$(NCURSES_DIR) --with-curses=$(CURSES_DIR) --with-termcap=$(TERMCAP_DIR) --with-tinfo=$(TINFO_DIR)
  	$(MAKE) -C editline libedit.a
diff --git a/net/asterisk/files/patch-main::manager.c b/net/asterisk/files/patch-main::manager.c
new file mode 100644
index 000000000000..686068b09e09
--- /dev/null
+++ b/net/asterisk/files/patch-main::manager.c
@@ -0,0 +1,31 @@
+--- main/manager.c.orig	2008-04-02 15:49:38.000000000 +0300
++++ main/manager.c	2008-04-02 15:42:18.000000000 +0300
+@@ -176,6 +176,7 @@
+ 	struct eventqent *eventq;
+ 	/* Timeout for ast_carefulwrite() */
+ 	int writetimeout;
++	int pending_event;
+ 	AST_LIST_ENTRY(mansession) list;
+ };
+ 
+@@ -2214,6 +2215,11 @@
+ 	fds[0].events = POLLIN;
+ 	do {
+ 		ast_mutex_lock(&s->__lock);
++		if (s->pending_event) {
++			s->pending_event = 0;
++			ast_mutex_unlock(&s->__lock);
++			return 0;
++		}
+ 		s->waiting_thread = pthread_self();
+ 		ast_mutex_unlock(&s->__lock);
+ 
+@@ -2475,6 +2481,8 @@
+ 		ast_mutex_lock(&s->__lock);
+ 		if (s->waiting_thread != AST_PTHREADT_NULL)
+ 			pthread_kill(s->waiting_thread, SIGURG);
++		else
++			s->pending_event = 1;
+ 		ast_mutex_unlock(&s->__lock);
+ 	}
+ 	AST_LIST_UNLOCK(&sessions);
diff --git a/net/asterisk/files/patch-res::res_features.c b/net/asterisk/files/patch-res::res_features.c
deleted file mode 100644
index c21775f1690d..000000000000
--- a/net/asterisk/files/patch-res::res_features.c
+++ /dev/null
@@ -1,144 +0,0 @@
---- res/res_features.c.orig	Wed Feb  6 16:55:29 2008
-+++ res/res_features.c	Wed Feb  6 17:02:00 2008
-@@ -477,18 +477,6 @@ int ast_masq_park_call(struct ast_channe
- }
- 
- 
--#define FEATURE_RETURN_HANGUP		-1
--#define FEATURE_RETURN_SUCCESSBREAK	 0
--#define FEATURE_RETURN_PBX_KEEPALIVE	AST_PBX_KEEPALIVE
--#define FEATURE_RETURN_NO_HANGUP_PEER	AST_PBX_NO_HANGUP_PEER
--#define FEATURE_RETURN_PASSDIGITS	 21
--#define FEATURE_RETURN_STOREDIGITS	 22
--#define FEATURE_RETURN_SUCCESS	 	 23
--#define FEATURE_RETURN_KEEPTRYING    24
--
--#define FEATURE_SENSE_CHAN	(1 << 0)
--#define FEATURE_SENSE_PEER	(1 << 1)
--
- /*! \brief
-  * set caller and callee according to the direction
-  */
-@@ -1052,32 +1040,35 @@ static int remap_feature(const char *nam
- 	return res;
- }
- 
--static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
-+void ast_features_lock(void)
-+{
-+	ast_rwlock_rdlock(&features_lock);
-+	AST_LIST_LOCK(&feature_list);
-+}
-+
-+void ast_features_unlock(void)
-+{
-+	AST_LIST_UNLOCK(&feature_list);
-+	ast_rwlock_unlock(&features_lock);
-+}
-+
-+int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result)
- {
- 	int x;
--	struct ast_flags features;
- 	int res = FEATURE_RETURN_PASSDIGITS;
- 	struct ast_call_feature *feature;
--	const char *dynamic_features;
-+	const char *dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
- 	char *tmp, *tok;
- 
--	if (sense == FEATURE_SENSE_CHAN) {
--		ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);	
--		dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
--	} else {
--		ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);	
--		dynamic_features = pbx_builtin_getvar_helper(peer, "DYNAMIC_FEATURES");
--	}
--	if (option_debug > 2)
--		ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d dynamic=%s\n", chan->name, peer->name, sense, features.flags, dynamic_features);
-+	result->builtin_feature = NULL;
-+	result->num_dyn_features = 0;
- 
--	ast_rwlock_rdlock(&features_lock);
- 	for (x = 0; x < FEATURES_COUNT; x++) {
--		if ((ast_test_flag(&features, builtin_features[x].feature_mask)) &&
-+		if ((ast_test_flag(features, builtin_features[x].feature_mask)) &&
- 		    !ast_strlen_zero(builtin_features[x].exten)) {
- 			/* Feature is up for consideration */
- 			if (!strcmp(builtin_features[x].exten, code)) {
--				res = builtin_features[x].operation(chan, peer, config, code, sense, NULL);
-+				result->builtin_feature = &builtin_features[x];
- 				break;
- 			} else if (!strncmp(builtin_features[x].exten, code, strlen(code))) {
- 				if (res == FEATURE_RETURN_PASSDIGITS)
-@@ -1085,7 +1076,6 @@ static int ast_feature_interpret(struct 
- 			}
- 		}
- 	}
--	ast_rwlock_unlock(&features_lock);
- 
- 	if (ast_strlen_zero(dynamic_features))
- 		return res;
-@@ -1093,9 +1083,7 @@ static int ast_feature_interpret(struct 
- 	tmp = ast_strdupa(dynamic_features);
- 
- 	while ((tok = strsep(&tmp, "#"))) {
--		AST_LIST_LOCK(&feature_list);	
- 		if (!(feature = find_dynamic_feature(tok))) {
--			AST_LIST_UNLOCK(&feature_list);
- 			continue;
- 		}
- 			
-@@ -1103,18 +1091,49 @@ static int ast_feature_interpret(struct 
- 		if (!strcmp(feature->exten, code)) {
- 			if (option_verbose > 2)
- 				ast_verbose(VERBOSE_PREFIX_3 " Feature Found: %s exten: %s\n",feature->sname, tok);
--			res = feature->operation(chan, peer, config, code, sense, feature);
--			if (res != FEATURE_RETURN_KEEPTRYING) {
--				AST_LIST_UNLOCK(&feature_list);
-+			result->dynamic_features[result->num_dyn_features++] = feature;
-+			if (result->num_dyn_features >= (sizeof(result->dynamic_features) / sizeof(result->dynamic_features[0]))) {
- 				break;
- 			}
- 			res = FEATURE_RETURN_PASSDIGITS;
- 		} else if (!strncmp(feature->exten, code, strlen(code)))
- 			res = FEATURE_RETURN_STOREDIGITS;
- 
--		AST_LIST_UNLOCK(&feature_list);
- 	}
- 	
-+	return res;
-+}
-+
-+static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
-+{
-+	struct feature_interpret_result result;
-+	int x;
-+	struct ast_flags features;
-+	int res = FEATURE_RETURN_PASSDIGITS;
-+	struct ast_call_feature *feature;
-+
-+	if (sense == FEATURE_SENSE_CHAN)
-+		ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);	
-+	else
-+		ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);	
-+	if (option_debug > 2)
-+		ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d\n", chan->name, peer->name, sense, features.flags);
-+
-+	ast_features_lock();
-+	res = ast_feature_detect(chan, &features, code, &result);
-+
-+	if (result.builtin_feature)
-+		res = result.builtin_feature->operation(chan, peer, config, code, sense, NULL);
-+
-+	for (x = 0; x < result.num_dyn_features; ++x) {
-+		feature = result.dynamic_features[x];
-+		res = feature->operation(chan, peer, config, code, sense, feature);
-+		if (res != FEATURE_RETURN_KEEPTRYING)
-+			break;
-+		res = FEATURE_RETURN_PASSDIGITS;
-+	}
-+
-+	ast_features_unlock();
- 	return res;
- }
- 
diff --git a/net/asterisk/files/rtp_force_dtmf-codecnego.diff b/net/asterisk/files/rtp_force_dtmf-codecnego.diff
new file mode 100644
index 000000000000..7cd43e08d238
--- /dev/null
+++ b/net/asterisk/files/rtp_force_dtmf-codecnego.diff
@@ -0,0 +1,86 @@
+--- channels/chan_sip.c.orig	2008-03-12 17:37:00.000000000 +0200
++++ channels/chan_sip.c	2008-03-12 18:17:33.000000000 +0200
+@@ -554,6 +554,9 @@
+ static unsigned int global_tos_sip;		/*!< IP type of service for SIP packets */
+ static unsigned int global_tos_audio;		/*!< IP type of service for audio RTP packets */
+ static unsigned int global_tos_video;		/*!< IP type of service for video RTP packets */
++static int global_force_dtmf_relay = 0;
++static int global_force_dtmf_relay_pt = 101;
++
+ static int compactheaders;		/*!< send compact sip headers */
+ static int recordhistory;		/*!< Record SIP history. Off by default */
+ static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
+@@ -4983,6 +4986,8 @@
+ 	int codec_index = 0;
+ 	int codec_pt_order[256];
+ 
++	int dtmf_present = 0;
++
+ 	if (!p->rtp) {
+ 		ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
+ 		return -1;
+@@ -5408,12 +5413,20 @@
+ 	for (x = 0; x < codec_index; ++x) {
+ 		struct rtpPayloadType pt;
+ 		pt = ast_rtp_lookup_pt(p->rtp, codec_pt_order[x]);
++		if (pt.code == AST_RTP_DTMF)
++			dtmf_present = 1;
+ 		if (!pt.isAstFormat && !pt.code && p->vrtp)
+ 			pt = ast_rtp_lookup_pt(p->vrtp, codec_pt_order[x]);
+ 		if (pt.isAstFormat)
+ 			ast_codec_pref_append(&p->formats, pt.code);
+ 	}
+ 	ast_codec_pref_remove2(&p->formats, ~p->usercapability);
++	if (!dtmf_present && global_force_dtmf_relay) {
++		newnoncodeccapability |= AST_RTP_DTMF;
++		ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
++		codec_pt_order[codec_index++] = global_force_dtmf_relay_pt;
++		ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
++	}
+ 
+ 	/* Now gather all of the codecs that we are asked for: */
+ 	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
+@@ -16845,6 +16858,9 @@
+ 
+ 	global_matchexterniplocally = FALSE;
+ 
++	global_force_dtmf_relay = 0;
++	global_force_dtmf_relay_pt = 101;
++
+ 	/* Copy the default jb config over global_jbconf */
+ 	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+ 
+@@ -16901,6 +16917,18 @@
+ 			}
+ 		} else if (!strcasecmp(v->name, "vmexten")) {
+ 			ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
++		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
++			if ((global_force_dtmf_relay = ast_true(v->value)))
++				ast_verbose("RTP DTMF relaying will be enforced\n");
++			else
++				ast_verbose("RTP DTMF relaying will not be enforced\n");
++		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
++			sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
++			if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
++				ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
++				global_force_dtmf_relay_pt = 101;
++			} else
++				ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
+ 		} else if (!strcasecmp(v->name, "rtptimeout")) {
+ 			if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
+ 				ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
+--- configs/sip.conf.sample.orig	2008-03-12 17:57:19.000000000 +0200
++++ configs/sip.conf.sample	2008-03-12 18:13:03.000000000 +0200
+@@ -53,6 +53,12 @@
+ 				; and multiline formatted headers for strict
+ 				; SIP compatibility (defaults to "no")
+ 
++;rtp_force_dtmf_relay=no	; Enable RFC2833 DTMFs to be sent even if peer 
++				; hasn't announced support for it. Default: no
++
++;rtp_force_dtmf_relay_pt=101	; RTP payload type value for enforced RFC2833
++				; DTMFs. Default: 101
++
+ ; See doc/README.tos for a description of these parameters.
+ ;tos_sip=cs3                    ; Sets TOS for SIP packets.
+ ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
diff --git a/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff
new file mode 100644
index 000000000000..776422b34645
--- /dev/null
+++ b/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff
@@ -0,0 +1,70 @@
+--- channels/chan_sip.c.orig	2008-01-31 21:52:49.000000000 +0200
++++ channels/chan_sip.c	2008-03-14 17:50:57.000000000 +0200
+@@ -556,6 +556,9 @@
+ static unsigned int global_tos_sip;		/*!< IP type of service for SIP packets */
+ static unsigned int global_tos_audio;		/*!< IP type of service for audio RTP packets */
+ static unsigned int global_tos_video;		/*!< IP type of service for video RTP packets */
++static int global_force_dtmf_relay = 0;
++static int global_force_dtmf_relay_pt = 101;
++
+ static int compactheaders;		/*!< send compact sip headers */
+ static int recordhistory;		/*!< Record SIP history. Off by default */
+ static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
+@@ -5392,6 +5395,13 @@
+ 
+ 	/* Now gather all of the codecs that we are asked for: */
+ 	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
++	/* Add telephone-event */
++	if (global_force_dtmf_relay && !(peernoncodeccapability & AST_RTP_DTMF)) {
++		ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
++		found_rtpmap_codecs[last_rtpmap_codec++] = global_force_dtmf_relay_pt;
++		ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
++		peernoncodeccapability |= AST_RTP_DTMF;
++	}
+ 	ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
+ 
+ 	newjointcapability = p->capability & (peercapability | vpeercapability);
+@@ -16833,6 +16843,9 @@
+ 
+ 	global_matchexterniplocally = FALSE;
+ 
++	global_force_dtmf_relay = 0;
++	global_force_dtmf_relay_pt = 101;
++
+ 	/* Copy the default jb config over global_jbconf */
+ 	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+ 
+@@ -16889,6 +16902,18 @@
+ 			}
+ 		} else if (!strcasecmp(v->name, "vmexten")) {
+ 			ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
++		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
++			if ((global_force_dtmf_relay = ast_true(v->value)))
++				ast_verbose("RTP DTMF relaying will be enforced\n");
++			else
++				ast_verbose("RTP DTMF relaying will not be enforced\n");
++		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
++			sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
++			if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
++				ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
++				global_force_dtmf_relay_pt = 101;
++			} else
++				ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
+ 		} else if (!strcasecmp(v->name, "rtptimeout")) {
+ 			if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
+ 				ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
+--- configs/sip.conf.sample.orig	2008-03-12 17:57:19.000000000 +0200
++++ configs/sip.conf.sample	2008-03-12 18:13:03.000000000 +0200
+@@ -53,6 +53,12 @@
+ 				; and multiline formatted headers for strict
+ 				; SIP compatibility (defaults to "no")
+ 
++;rtp_force_dtmf_relay=no	; Enable RFC2833 DTMFs to be sent even if peer 
++				; hasn't announced support for it. Default: no
++
++;rtp_force_dtmf_relay_pt=101	; RTP payload type value for enforced RFC2833
++				; DTMFs. Default: 101
++
+ ; See doc/README.tos for a description of these parameters.
+ ;tos_sip=cs3                    ; Sets TOS for SIP packets.
+ ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
diff --git a/net/asterisk/files/sip_force_callid.diff b/net/asterisk/files/sip_force_callid.diff
new file mode 100644
index 000000000000..67a3e0192633
--- /dev/null
+++ b/net/asterisk/files/sip_force_callid.diff
@@ -0,0 +1,12 @@
+--- channels/chan_sip.c.orig	2008-04-09 11:39:09.000000000 +0300
++++ channels/chan_sip.c	2008-04-09 11:40:28.000000000 +0300
+@@ -2998,6 +2998,9 @@
+ 			p->t38.state = T38_LOCAL_DIRECT;
+ 			if (option_debug)
+ 				ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
++		} else if (!strcasecmp(ast_var_name(current), "SIP_FORCE_CALLID")) {
++			
++			ast_string_field_set(p, callid, ast_var_value(current));
+ 		}
+ 
+ 	}
diff --git a/net/asterisk/pkg-plist b/net/asterisk/pkg-plist
index 7bcd9f5265f4..85ea7e48e505 100644
--- a/net/asterisk/pkg-plist
+++ b/net/asterisk/pkg-plist
@@ -295,7 +295,6 @@ lib/asterisk/modules/codec_adpcm.so
 lib/asterisk/modules/codec_alaw.so
 lib/asterisk/modules/codec_g726.so
 lib/asterisk/modules/codec_gsm.so
-lib/asterisk/modules/codec_ilbc.so
 lib/asterisk/modules/codec_lpc10.so
 lib/asterisk/modules/codec_speex.so
 lib/asterisk/modules/codec_ulaw.so
@@ -380,8 +379,9 @@ share/asterisk/moh/LICENSE-asterisk-moh-freeplay-wav
 share/asterisk/moh/fpm-calm-river.wav
 share/asterisk/moh/fpm-sunshine.wav
 share/asterisk/moh/fpm-world-mix.wav
-share/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.8
-share/asterisk/sounds/CREDITS-asterisk-core-en-1.4.8
+share/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.9
+share/asterisk/sounds/CHANGES-asterisk-core-en-1.4.9
+share/asterisk/sounds/CREDITS-asterisk-core-en-1.4.9
 share/asterisk/sounds/agent-alreadyon.gsm
 share/asterisk/sounds/agent-incorrect.gsm
 share/asterisk/sounds/agent-loggedoff.gsm
@@ -526,15 +526,19 @@ share/asterisk/sounds/digits/thousand.gsm
 share/asterisk/sounds/digits/today.gsm
 share/asterisk/sounds/digits/tomorrow.gsm
 share/asterisk/sounds/digits/yesterday.gsm
+share/asterisk/sounds/dir-first.gsm
+share/asterisk/sounds/dir-firstlast.gsm
 share/asterisk/sounds/dir-instr.gsm
 share/asterisk/sounds/dir-intro-fn.gsm
 share/asterisk/sounds/dir-intro.gsm
+share/asterisk/sounds/dir-last.gsm
 share/asterisk/sounds/dir-multi1.gsm
 share/asterisk/sounds/dir-multi2.gsm
 share/asterisk/sounds/dir-multi3.gsm
 share/asterisk/sounds/dir-multi9.gsm
 share/asterisk/sounds/dir-nomatch.gsm
 share/asterisk/sounds/dir-nomore.gsm
+share/asterisk/sounds/dir-pls-enter.gsm
 share/asterisk/sounds/followme/call-from.gsm
 share/asterisk/sounds/followme/no-recording.gsm
 share/asterisk/sounds/followme/options.gsm
@@ -669,12 +673,20 @@ share/asterisk/sounds/silence/7.gsm
 share/asterisk/sounds/silence/8.gsm
 share/asterisk/sounds/silence/9.gsm
 share/asterisk/sounds/spy-agent.gsm
+share/asterisk/sounds/spy-console.gsm
 share/asterisk/sounds/spy-h323.gsm
 share/asterisk/sounds/spy-iax.gsm
 share/asterisk/sounds/spy-iax2.gsm
+share/asterisk/sounds/spy-jingle.gsm
+share/asterisk/sounds/spy-local.gsm
 share/asterisk/sounds/spy-mgcp.gsm
+share/asterisk/sounds/spy-misdn.gsm
+share/asterisk/sounds/spy-mobile.gsm
+share/asterisk/sounds/spy-nbs.gsm
 share/asterisk/sounds/spy-sip.gsm
 share/asterisk/sounds/spy-skinny.gsm
+share/asterisk/sounds/spy-unistim.gsm
+share/asterisk/sounds/spy-usbradio.gsm
 share/asterisk/sounds/spy-zap.gsm
 share/asterisk/sounds/ss-noservice.gsm
 share/asterisk/sounds/transfer.gsm
@@ -716,6 +728,7 @@ share/asterisk/sounds/vm-incorrect-mailbox.gsm
 share/asterisk/sounds/vm-incorrect.gsm
 share/asterisk/sounds/vm-instructions.gsm
 share/asterisk/sounds/vm-intro.gsm
+share/asterisk/sounds/vm-invalidpassword.gsm
 share/asterisk/sounds/vm-isonphone.gsm
 share/asterisk/sounds/vm-isunavail.gsm
 share/asterisk/sounds/vm-last.gsm
@@ -738,8 +751,10 @@ share/asterisk/sounds/vm-nomore.gsm
 share/asterisk/sounds/vm-nonumber.gsm
 share/asterisk/sounds/vm-num-i-have.gsm
 share/asterisk/sounds/vm-onefor.gsm
+share/asterisk/sounds/vm-onefor-full.gsm
 share/asterisk/sounds/vm-options.gsm
 share/asterisk/sounds/vm-opts.gsm
+share/asterisk/sounds/vm-opts-full.gsm
 share/asterisk/sounds/vm-passchanged.gsm
 share/asterisk/sounds/vm-password.gsm
 share/asterisk/sounds/vm-press.gsm
@@ -767,6 +782,7 @@ share/asterisk/sounds/vm-tempgreeting2.gsm
 share/asterisk/sounds/vm-tempremoved.gsm
 share/asterisk/sounds/vm-then-pound.gsm
 share/asterisk/sounds/vm-theperson.gsm
+share/asterisk/sounds/vm-tmpexists.gsm
 share/asterisk/sounds/vm-tocallback.gsm
 share/asterisk/sounds/vm-tocallnum.gsm
 share/asterisk/sounds/vm-tocancel.gsm