svn commit: r382011 - in head: . net/asterisk13 net/pjsip
Guido Falsi
madpilot at FreeBSD.org
Mon Mar 23 15:46:25 UTC 2015
Author: madpilot
Date: Mon Mar 23 15:46:23 2015
New Revision: 382011
URL: https://svnweb.freebsd.org/changeset/ports/382011
QAT: https://qat.redports.org/buildarchive/r382011/
Log:
Due to recent changes in OpenSSL, disable SRTP support for asterisk13
by default and avoid pjsip pulling in libsrtp, otherwise a not
working package would be generated.
Add note to UPDATING to keep users informed.
Modified:
head/UPDATING
head/net/asterisk13/Makefile
head/net/pjsip/Makefile
head/net/pjsip/pkg-plist
Modified: head/UPDATING
==============================================================================
--- head/UPDATING Mon Mar 23 15:28:14 2015 (r382010)
+++ head/UPDATING Mon Mar 23 15:46:23 2015 (r382011)
@@ -5,6 +5,20 @@ they are unavoidable.
You should get into the habit of checking this file for changes each time
you update your ports collection, before attempting any port upgrades.
+20150323:
+ AFFECTS: Users of net/asterisk* and net/pjsip ports
+ AUTHOR: madpilot at FreeBSD.org
+
+ Due to conflicts between base OpenSSL and ports provided OpenSSL
+ library, which is required by net/libsrtp, the srtp support has
+ to be removed from the default asterisk13 port configuration,
+ otherwise a a not working binary would be generated.
+
+ To get SRTP support working in the asterisk ports it is needed
+ to build all other required ports with WITH_OPENSSL_PORT=yes. For
+ asterisk13 it is also needed to enable the EXTSRTP option in the
+ pjsip port.
+
20150322:
AFFECTS: Users of security/openssh-portable
AUTHOR: bdrewery at FreeBSD.org
Modified: head/net/asterisk13/Makefile
==============================================================================
--- head/net/asterisk13/Makefile Mon Mar 23 15:28:14 2015 (r382010)
+++ head/net/asterisk13/Makefile Mon Mar 23 15:46:23 2015 (r382011)
@@ -2,6 +2,7 @@
PORTNAME= asterisk
PORTVERSION= 13.2.0
+PORTREVISION= 1
CATEGORIES= net
MASTER_SITES= http://downloads.asterisk.org/pub/telephony/asterisk/ \
http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -47,7 +48,7 @@ CONFLICTS_INSTALL= asterisk*-1.8* asteri
OPTIONS_DEFINE= VORBIS PGSQL MYSQL ODBC RADIUS SNMP FREETDS XMPP SQLITE GSM \
CURL SPANDSP EXCHANGE NEWG711 SRTP LUA LDAP OOH323 PJSIP SPEEX
OPTIONS_DEFAULT= VORBIS ODBC PGSQL RADIUS SNMP FREETDS \
- XMPP GSM SQLITE3 CURL SRTP LUA PJSIP SPEEX
+ XMPP GSM SQLITE3 CURL LUA PJSIP SPEEX
OPTIONS_DEFINE_i386= DAHDI
OPTIONS_DEFINE_amd64= DAHDI
@@ -59,7 +60,7 @@ OPTIONS_DEFAULT_sparc64= DAHDI
EXCHANGE_DESC?= Exchange calendar support
NEWG711_DESC?= New G711 Codec
-SRTP_DESC?= SecureRTP support
+SRTP_DESC?= SecureRTP support (Needs all ports build with WITH_OPENSSL_PORT=yes)
OOH323_DESC?= ooh323 support
DAHDI_DESC?= DAHDI support
XMPP_DESC?= XMPP/GTALK support
Modified: head/net/pjsip/Makefile
==============================================================================
--- head/net/pjsip/Makefile Mon Mar 23 15:28:14 2015 (r382010)
+++ head/net/pjsip/Makefile Mon Mar 23 15:46:23 2015 (r382011)
@@ -2,7 +2,7 @@
PORTNAME= pjsip
PORTVERSION= 2.3
-PORTREVISION= 1
+PORTREVISION= 2
CATEGORIES= net
MASTER_SITES= http://www.pjsip.org/release/${PORTVERSION}/
DISTNAME= pjproject-${DISTVERSION}
@@ -12,12 +12,10 @@ COMMENT= Multimedia communication librar
LICENSE= GPLv2
-LIB_DEPENDS= libportaudio.so.2:${PORTSDIR}/audio/portaudio2 \
- libsrtp.so:${PORTSDIR}/net/libsrtp
+LIB_DEPENDS= libportaudio.so.2:${PORTSDIR}/audio/portaudio2
GNU_CONFIGURE= yes
CONFIGURE_ARGS= --with-external-pa \
- --with-external-srtp \
--disable-silk
USES= gmake tar:bz2
USE_LDCONFIG= yes
@@ -27,7 +25,7 @@ CFLAGS+= -I${LOCALBASE}/include/portaudi
MAKE_JOBS_UNSAFE= yes
OPTIONS_DEFINE= FFMPEG G711 G722 G7221 GSM ILBC L16 OPENH264 RESAMPLE RESAMPLEDLL \
- SAMPLERATE SDL SHARED SPEEX SPEEXAEC V4L IPV6 SOUND VIDEO AMR
+ SAMPLERATE SDL SHARED SPEEX SPEEXAEC V4L IPV6 SOUND VIDEO AMR EXTSRTP
OPTIONS_DEFAULT=G711 G722 G7221 GSM ILBC L16 SHARED SPEEX SPEEXAEC
G711_DESC= G.711 codec support
@@ -42,6 +40,7 @@ SHARED_DESC= Build shared libraries (oth
SPEEXAEC_DESC= Speex Acoustic Echo Canceller/AEC
V4L_DESC= Video4Linux2 support
YUV_DESC= Libyuv support
+EXTSRTP_DESC= Use libsrtp port (needs all ports compiled with WITH_OPENSSL_PORT=yes)
OPTIONS_SUB= yes
@@ -77,6 +76,8 @@ VIDEO_CONFIGURE_ENABLE= video
AMR_CONFIGURE_WITH= opencore-amr
AMR_LIB_DEPENDS= libopencore-amrwb.so:${PORTSDIR}/audio/opencore-amr \
libvo-amrwbenc.so:${PORTSDIR}/audio/vo-amrwbenc
+EXTSRTP_CONFIGURE_WITH= external-srtp
+EXTSRTP_LIB_DEPENDS= libsrtp.so:${PORTSDIR}/net/libsrtp
post-patch:
@${REINPLACE_CMD} -e 's|%%LOCALBASE%%|${LOCALBASE}|' \
Modified: head/net/pjsip/pkg-plist
==============================================================================
--- head/net/pjsip/pkg-plist Mon Mar 23 15:28:14 2015 (r382010)
+++ head/net/pjsip/pkg-plist Mon Mar 23 15:46:23 2015 (r382011)
@@ -342,4 +342,7 @@ lib/libpjsua2-%%CONFIGURE_TARGET%%.a
%%NO_SAMPLERATE%%%%RESAMPLE%%lib/libresample-%%CONFIGURE_TARGET%%.a
%%NO_SAMPLERATE%%%%RESAMPLE%%%%RESAMPLEDLL%%lib/libresample.so
%%NO_SAMPLERATE%%%%RESAMPLE%%%%RESAMPLEDLL%%lib/libresample.so.2
+%%NO_EXTSRTP%%lib/libsrtp-%%CONFIGURE_TARGET%%.a
+%%NO_EXTSRTP%%%%SHARED%%lib/libsrtp.so
+%%NO_EXTSRTP%%%%SHARED%%lib/libsrtp.so.2
libdata/pkgconfig/libpjproject.pc
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