kpriv/freebsd-ports
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity

Update to 1.2.3.

Browse source
This commit is contained in:
Maxim Sobolev 2006-01-27 00:15:33 +00:00
parent 5a7bdcd5ba
commit f644d74fb3
Notes: svn2git 2021-03-31 03:12:20 +00:00 
svn path=/head/; revision=154549
49 changed files with 343 additions and 266 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

5
net/asterisk-bristuff/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk-bristuff/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk-bristuff/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk-bristuff/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk-bristuff/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk-bristuff/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk-bristuff/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk-devel/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk-devel/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk-devel/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk-devel/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk-devel/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk-devel/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk-devel/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk10/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk10/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk10/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk10/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk10/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk10/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk10/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk12/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk12/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk12/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk12/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk12/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk12/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk12/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk14/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk14/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk14/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk14/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk14/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk14/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk14/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;

5
net/asterisk16/Makefile
View file

@ -6,7 +6,7 @@
#
PORTNAME= asterisk
PORTVERSION= 1.2.1
PORTVERSION= 1.2.3
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/ \
@ -16,6 +16,9 @@ MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
MAINTAINER= sobomax@FreeBSD.org
COMMENT= An Open Source PBX and telephony toolkit
PATCHFILES= asterisk122_codec_negotiation-20060125.diff.gz
PATCH_SITES= http://www.portaone.com/~bamby/public/
BUILD_DEPENDS= mpg123:${PORTSDIR}/audio/mpg123
LIB_DEPENDS= speex.3:${PORTSDIR}/audio/speex \
newt.51:${PORTSDIR}/devel/newt \

9
net/asterisk16/distinfo
View file

@ -1,3 +1,6 @@
MD5 (asterisk-1.2.1.tar.gz) = 04657086791e80f319c0d728af705001
SHA256 (asterisk-1.2.1.tar.gz) = f0d3ed36353c1f5726225d2794bea7d1d1a92c9e58336b02b293f43dd4b97948
SIZE (asterisk-1.2.1.tar.gz) = 10517194
MD5 (asterisk-1.2.3.tar.gz) = 89d90096f6c7acf7d3dfe5457a2e3dbe
SHA256 (asterisk-1.2.3.tar.gz) = e305c1dd3e875f22626913812851a344dd89699af545626aa5553bc4f1e1b943
SIZE (asterisk-1.2.3.tar.gz) = 10533725
MD5 (asterisk122_codec_negotiation-20060125.diff.gz) = 165ca0d6eb9dc1205f24778d830e9713
SHA256 (asterisk122_codec_negotiation-20060125.diff.gz) = 8a337b84d0ec75130ae2f39660f4a1a9a762277ab6929fe4b7be22e675fd0ae1
SIZE (asterisk122_codec_negotiation-20060125.diff.gz) = 27944

20
net/asterisk16/files/patch-Makefile
View file

@ -68,7 +68,7 @@ $FreeBSD$
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
@@ -218,13 +218,13 @@
@@ -218,25 +218,25 @@
endif
INCLUDE+=-Iinclude -I../include
@ -85,7 +85,11 @@ $FreeBSD$
endif
ifneq ($(PROC),ultrasparc)
@@ -235,8 +235,8 @@
- ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+# ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
@ -455,10 +459,10 @@ $FreeBSD$
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
@@ -759,25 +737,25 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -800,25 +778,25 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
- ) > $(DESTDIR)$(ASTCONFPATH) ; \
+ ) > $(DESTDIR)$(ASTCONFPATH)-dist ; \
else \
@ -487,7 +491,7 @@ $FreeBSD$
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
@@ -791,7 +769,7 @@
@@ -832,7 +810,7 @@
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
@ -496,7 +500,7 @@ $FreeBSD$
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@@ -820,10 +798,10 @@
@@ -861,10 +839,10 @@
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \

14
net/asterisk16/files/patch-channels::chan_h323.c
View file

@ -1,14 +0,0 @@
$FreeBSD$
--- channels/chan_h323.c.orig Tue Nov 29 20:24:39 2005
+++ channels/chan_h323.c Thu Jan 12 18:29:50 2006
@@ -2281,7 +2281,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;

20
net/asterisk16/files/patch-channels::chan_sip.c
View file

@ -1,9 +1,9 @@
$FreeBSD$
--- channels/chan_sip.c.orig Thu Jan 12 18:43:37 2006
+++ channels/chan_sip.c Thu Jan 12 18:50:11 2006
@@ -336,7 +336,7 @@ static char global_vmexten[AST_MAX_EXTEN
--- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
static char default_language[MAX_LANGUAGE] = "";
@ -12,7 +12,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -474,6 +474,7 @@ struct sip_invite_param {
@@ -475,6 +475,7 @@ struct sip_invite_param {
struct sip_route {
struct sip_route *next;
@ -20,7 +20,7 @@ $FreeBSD$
char hop[0];
};
@@ -5993,6 +5994,7 @@ static void build_route(struct sip_pvt *
@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -28,7 +28,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6018,31 +6020,41 @@ static void build_route(struct sip_pvt *
@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -94,7 +94,7 @@ $FreeBSD$
}
}
}
@@ -10336,7 +10348,11 @@ static int handle_request_invite(struct
@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -107,7 +107,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
@@ -10364,7 +10380,6 @@ static int handle_request_invite(struct
@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@ -115,15 +115,15 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10450,6 +10465,13 @@ static int handle_request_invite(struct
@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);
+ } else {
+ p->jointcapability = p->capability;
+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
+ }
transmit_response_with_sdp(p, "200 OK", req, 1);

5
net/asterisk16/files/patch-codecs::gsm::Makefile
View file

@ -1,14 +1,15 @@
$FreeBSD$
--- codecs/gsm/Makefile
--- codecs/gsm/Makefile.orig
+++ codecs/gsm/Makefile
@@ -49,7 +49,9 @@
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
ifneq (${PROC},ppc64)
-OPTIMIZE+=-march=$(PROC)
+ifneq (${PROC},amd64)
OPTIMIZE+=-march=$(PROC)
+#OPTIMIZE+=-march=$(PROC)
+endif
endif
endif

14
net/asterisk16/files/patch-frame.c Normal file
View file

@ -0,0 +1,14 @@
$FreeBSD$
--- frame.c.orig Wed Jan 25 13:22:43 2006
+++ frame.c Wed Jan 25 13:21:25 2006
@@ -1170,7 +1170,7 @@ static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
- return 0;
+ return 2;
break;
case TYPE_SILENCE:
return 4;