svn commit: r382011 - in head: . net/asterisk13 net/pjsip

[Guido Falsi]

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