PKG_NAME:=pjproject
PKG_VERSION:=2.6
-PKG_RELEASE:=2
+PKG_RELEASE:=3
PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)/
CATEGORY:=Libraries
SUBMENU:=Telephony
URL:=http://www.pjsip.org/
- DEPENDS:=+libuuid +libstdcpp +libpthread
+ DEPENDS:=+libopenssl +libuuid +libstdcpp +libpthread
endef
define Package/pjproject/install/lib
endef
CONFIGURE_ARGS += \
- --enable-shared \
- --disable-floating-point \
- --enable-g711-codec \
- --disable-l16-codec \
+ $(if $(CONFIG_SOFT_FLOAT),--disable-floating-point) \
+ --disable-ext-sound \
+ --disable-ffmpeg \
+ --disable-g711-codec \
--disable-g722-codec \
--disable-g7221-codec \
--disable-gsm-codec \
- --disable-ilbc-coder \
+ --disable-ilbc-codec \
--disable-ipp \
- --disable-ssl \
- --disable-oss \
- --disable-sound \
+ --disable-l16-codec \
--disable-libwebrtc \
--disable-libyuv \
- --with-external-srtp="$(STAGING_DIR)/usr" \
- --without-external-gsm \
- --disable-small-filter \
- --disable-large-filter \
- --disable-speex-aec \
- --disable-g711-codec \
- --disable-l16-codec \
- --disable-gsm-codec \
- --disable-g722-codec \
- --disable-g7221-codec \
- --disable-speex-codec \
- --disable-ilbc-codec \
+ --disable-opencore-amr \
+ --disable-openh264 \
--disable-opus \
- --disable-resample-dll \
+ --disable-oss \
+ --disable-resample \
--disable-sdl \
- --disable-ffmpeg \
- --disable-v4l2
+ --disable-silk \
+ --disable-sound \
+ --disable-speex-aec \
+ --disable-speex-codec \
+ --disable-v4l2 \
+ --disable-video \
+ --enable-shared \
+ --with-external-srtp="$(STAGING_DIR)/usr" \
+ --with-ssl="$(STAGING_DIR)/usr" \
+ --without-external-gsm \
+ --without-external-pa \
+ --without-external-webrtc
TARGET_LDFLAGS+=-lc $(LIBGCC) -lm
TARGET_CFLAGS+=$(EXTRA_CFLAGS) $(TARGET_CPPFLAGS) $(EXTRA_CPPFLAGS)
PJPROJECT_LIBS = \
libpj libpjlib-util libpjmedia libpjnath libpjsip-simple \
- libpjsip-ua libpjsip libpjsua libpjsua2 libresample
+ libpjsip-ua libpjsip libpjsua libpjsua2
define Build/InstallDev
$(INSTALL_DIR) $(1)/usr/{include,lib}
$(eval $(call PJSIPpackage,libpj,libpj,+librt))
$(eval $(call PJSIPpackage,libpjlib-util,libpjlib-util,+libpj +librt))
-$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +libresample +librt +libspeex +libsrtp))
+$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +librt +libsrtp))
$(eval $(call PJSIPpackage,libpjnath,libpjnath,+libpj +libpjlib-util +librt))
-$(eval $(call PJSIPpackage,libpjsip-simple,libpjsip-simple,+libpj +libpjlib-util +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsip-ua,libpjsip-ua,+libpj +libpjlib-util +libpjmedia +libpjsip-simple +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsua,libpjsua,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsua2,libpjsua2,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libresample +librt +libspeex +libsrtp +libpjsua))
-$(eval $(call PJSIPpackage,libresample,libresample,))
+$(eval $(call PJSIPpackage,libpjsip-simple,libpjsip-simple,+libpj +libpjlib-util +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsip-ua,libpjsip-ua,+libpj +libpjlib-util +libpjmedia +libpjsip-simple +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +librt +libsrtp))
+$(eval $(call PJSIPpackage,libpjsua,libpjsua,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsua2,libpjsua2,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt +libpjsua))
--- /dev/null
+--- /dev/null
++++ b/pjlib/include/pj/config_site.h
+@@ -0,0 +1,92 @@
++/*
++ * Asterisk config_site.h
++ */
++
++#include <sys/select.h>
++
++/*
++ * Since both pjproject and asterisk source files will include config_site.h,
++ * we need to make sure that only pjproject source files include asterisk_malloc_debug.h.
++ */
++
++/* #if defined(MALLOC_DEBUG) && !defined(_ASTERISK_ASTMM_H)
++ * #include "asterisk_malloc_debug.h"
++ * #endif
++ */
++
++/*
++ * Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
++ * It only disables the pjmedia srtp transport which Asterisk doesn't use.
++ * The reason for the disable is that while Asterisk works fine with older libsrtp
++ * versions, newer versions of pjproject won't compile with them.
++ */
++
++/*
++ * This doesn't disable SRTP completely, so we have to keep using the external
++ * libsrtp, otherwise pjsip would just build the internal one.
++ */
++
++#define PJMEDIA_HAS_SRTP 0
++
++/*
++ * Defining PJMEDIA_HAS_WEBRTC_AEC to 0 does NOT disable Asterisk's ability to use
++ * webrtc. It only disables the pjmedia webrtc transport which Asterisk doesn't use.
++ */
++#define PJMEDIA_HAS_WEBRTC_AEC 0
++
++#define PJ_HAS_IPV6 1
++#define NDEBUG 1
++#define PJ_MAX_HOSTNAME (256)
++#define PJSIP_MAX_URL_SIZE (512)
++#ifdef PJ_HAS_LINUX_EPOLL
++#define PJ_IOQUEUE_MAX_HANDLES (5000)
++#else
++#define PJ_IOQUEUE_MAX_HANDLES (FD_SETSIZE)
++#endif
++#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
++#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
++
++#define PJ_SCANNER_USE_BITWISE 0
++#define PJ_OS_HAS_CHECK_STACK 0
++
++#ifndef PJ_LOG_MAX_LEVEL
++#define PJ_LOG_MAX_LEVEL 6
++#endif
++
++#define PJ_ENABLE_EXTRA_CHECK 1
++#define PJSIP_MAX_TSX_COUNT ((64*1024)-1)
++#define PJSIP_MAX_DIALOG_COUNT ((64*1024)-1)
++#define PJSIP_UDP_SO_SNDBUF_SIZE (512*1024)
++#define PJSIP_UDP_SO_RCVBUF_SIZE (512*1024)
++#define PJ_DEBUG 0
++#define PJSIP_SAFE_MODULE 0
++#define PJ_HAS_STRICMP_ALNUM 0
++
++/*
++ * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
++ * inconsistently used when calculating the hash value and doesn't
++ * convert the same characters as pj_tolower()/tolower(). Thus you
++ * can get different hash values if the string hashed has certain
++ * characters in it. (ASCII '@', '[', '\\', ']', '^', and '_')
++ */
++#undef PJ_HASH_USE_OWN_TOLOWER
++
++/*
++ It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
++ Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
++*/
++#undef PJSIP_UNESCAPE_IN_PLACE
++#define PJSIP_MAX_PKT_LEN 6000
++
++#undef PJ_TODO
++#define PJ_TODO(x)
++
++/* Defaults too low for WebRTC */
++#define PJ_ICE_MAX_CAND 32
++#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
++
++/* Increase limits to allow more formats */
++#define PJMEDIA_MAX_SDP_FMT 64
++#define PJMEDIA_MAX_SDP_BANDW 4
++#define PJMEDIA_MAX_SDP_ATTR (PJMEDIA_MAX_SDP_FMT*2 + 4)
++#define PJMEDIA_MAX_SDP_MEDIA 16
PKG_NAME:=asterisk13
PKG_VERSION:=13.17.0
-PKG_RELEASE:=2
+PKG_RELEASE:=3
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=http://downloads.asterisk.org/pub/telephony/asterisk/releases/
$(eval $(call BuildAsterisk13Module,pbx-realtime,Realtime Switch,realtime switch support,,,pbx_realtime,,))
$(eval $(call BuildAsterisk13Module,pbx-spool,Call Spool,outgoing call spool support,,,pbx_spool,,))
$(eval $(call BuildAsterisk13Module,pgsql,PostgreSQL,PostgreSQL support,+libpq @!arc,cel_pgsql.conf cdr_pgsql.conf res_pgsql.conf,cel_pgsql cdr_pgsql res_config_pgsql,,))
-$(eval $(call BuildAsterisk13Module,pjsip,pjsip channel,the channel pjsip,+asterisk13-res-sorcery +asterisk13-res-pjproject +libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,pjsip.conf pjsip_notify.conf,func_pjsip_endpoint chan_pjsip res_pjsip_acl res_pjsip_authenticator_digest res_pjsip_caller_id res_pjsip_dialog_info_body_generator res_pjsip_diversion res_pjsip_dtmf_info res_pjsip_endpoint_identifier_anonymous res_pjsip_endpoint_identifier_ip res_pjsip_endpoint_identifier_user res_pjsip_exten_state res_pjsip_header_funcs res_pjsip_log_forwarder res_pjsip_logger res_pjsip_messaging res_pjsip_multihomed res_pjsip_mwi_body_generator res_pjsip_mwi res_pjsip_nat res_pjsip_notify res_pjsip_one_touch_record_info res_pjsip_outbound_authenticator_digest res_pjsip_outbound_publish res_pjsip_outbound_registration res_pjsip_path res_pjsip_pidf_body_generator res_pjsip_pidf_digium_body_supplement res_pjsip_pidf_eyebeam_body_supplement res_pjsip_publish_asterisk res_pjsip_pubsub res_pjsip_refer res_pjsip_registrar_expire res_pjsip_registrar res_pjsip_rfc3326 res_pjsip_sdp_rtp res_pjsip_send_to_voicemail res_pjsip_session res_pjsip res_pjsip_transport_websocket res_pjsip_t38 res_pjsip_xpidf_body_generator,,))
+$(eval $(call BuildAsterisk13Module,pjsip,pjsip channel,the channel pjsip,+asterisk13-res-sorcery +asterisk13-res-pjproject +libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,pjsip.conf pjsip_notify.conf,chan_pjsip func_pjsip_aor func_pjsip_contact func_pjsip_endpoint res_pjsip res_pjsip_acl res_pjsip_authenticator_digest res_pjsip_caller_id res_pjsip_config_wizard res_pjsip_dialog_info_body_generator res_pjsip_diversion res_pjsip_dlg_options res_pjsip_dtmf_info res_pjsip_empty_info res_pjsip_endpoint_identifier_anonymous res_pjsip_endpoint_identifier_ip res_pjsip_endpoint_identifier_user res_pjsip_exten_state res_pjsip_header_funcs res_pjsip_history res_pjsip_log_forwarder res_pjsip_logger res_pjsip_messaging res_pjsip_multihomed res_pjsip_mwi res_pjsip_mwi_body_generator res_pjsip_nat res_pjsip_notify res_pjsip_one_touch_record_info res_pjsip_outbound_authenticator_digest res_pjsip_outbound_publish res_pjsip_outbound_registration res_pjsip_path res_pjsip_pidf_body_generator res_pjsip_pidf_digium_body_supplement res_pjsip_pidf_eyebeam_body_supplement res_pjsip_publish_asterisk res_pjsip_pubsub res_pjsip_refer res_pjsip_registrar res_pjsip_registrar_expire res_pjsip_rfc3326 res_pjsip_sdp_rtp res_pjsip_send_to_voicemail res_pjsip_session res_pjsip_sips_contact res_pjsip_t38 res_pjsip_transport_management res_pjsip_transport_websocket res_pjsip_xpidf_body_generator,,))
$(eval $(call BuildAsterisk13Module,res-adsi,Provide ADSI,Analog Display Services Interface capability,,,res_adsi,,))
$(eval $(call BuildAsterisk13Module,res-ael-share,Shareable AEL code,support for shareable AEL code mainly between internal and external modules,,,res_ael_share,,))
$(eval $(call BuildAsterisk13Module,res-agi,Asterisk Gateway Interface,Support for the Asterisk Gateway Interface extension,+asterisk13-res-speech,,res_agi,,))
$(eval $(call BuildAsterisk13Module,res-musiconhold,MOH,Music On Hold support,,musiconhold.conf,res_musiconhold,,))
$(eval $(call BuildAsterisk13Module,res-parking,Phone Parking,Phone Parking application,,res_parking.conf,res_parking,,))
$(eval $(call BuildAsterisk13Module,res-phoneprov,Phone Provisioning,Phone provisioning application for the asterisk internal http server,,phoneprov.conf,res_phoneprov,,))
-$(eval $(call BuildAsterisk13Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libresample +libsrtp,pjproject.conf,res_pjproject,,))
+$(eval $(call BuildAsterisk13Module,res-pjsip-phoneprov,PJSIP Phone Provisioning,PJSIP Phone Provisioning,+asterisk13-pjsip +asterisk13-res-phoneprov,,res_pjsip_phoneprov_provider,,))
+$(eval $(call BuildAsterisk13Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp,pjproject.conf,res_pjproject,,))
$(eval $(call BuildAsterisk13Module,res-realtime,RealTime CLI,RealTime CLI,,,res_realtime,,))
$(eval $(call BuildAsterisk13Module,res-rtp-asterisk,RTP stack,Supports RTP and RTCP with Symmetric RTP support for NAT traversal,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,rtp.conf,res_rtp_asterisk,,))
$(eval $(call BuildAsterisk13Module,res-rtp-multicast,RTP multicast engine,Multicast RTP Engine,,,res_rtp_multicast,,))
--- /dev/null
+Description: Fix unresolved symbol ast_sip_session_unregister_supplement in PJSIP
+Origin: other, https://issues.asterisk.org/jira/secure/attachment/54695/ast.diff
+Bug: https://issues.asterisk.org/jira/browse/ASTERISK-26518
+
+--- a/res/res_pjsip/pjsip_message_ip_updater.c
++++ b/res/res_pjsip/pjsip_message_ip_updater.c
+@@ -362,15 +362,15 @@ void ast_res_pjsip_cleanup_message_ip_up
+ {
+ ast_sip_unregister_service(&multihomed_module);
+ ast_sip_unregister_supplement(&multihomed_supplement);
+- ast_sip_session_unregister_supplement(&multihomed_session_supplement);
++ //ast_sip_session_unregister_supplement(&multihomed_session_supplement);
+ }
+
+ int ast_res_pjsip_init_message_ip_updater(void)
+ {
+- if (ast_sip_session_register_supplement(&multihomed_session_supplement)) {
++ /*if (ast_sip_session_register_supplement(&multihomed_session_supplement)) {
+ ast_log(LOG_ERROR, "Could not register multihomed session supplement for outgoing requests\n");
+ return -1;
+- }
++ }*/
+
+ if (ast_sip_register_supplement(&multihomed_supplement)) {
+ ast_log(LOG_ERROR, "Could not register multihomed supplement for outgoing requests\n");
PKG_NAME:=asterisk-g72x
PKG_VERSION:=1.4.1
-PKG_RELEASE:=1
+PKG_RELEASE:=2
PKG_SOURCE:=asterisk-g72x-$(PKG_VERSION).tar.bz2
PKG_SOURCE_URL:=http://asterisk.hosting.lv/src/
PKG_LICENSE_FILES:=README.md
PKG_MAINTAINER:=Alex Samorukov <samm@os2.kiev.ua>
+PKG_BUILD_DIR:=$(BUILD_DIR)/$(PKG_NAME)-$(BUILD_VARIANT)/$(PKG_NAME)-$(PKG_VERSION)
+
include $(INCLUDE_DIR)/package.mk
define Package/asterisk-g72x/Default
#
-# Copyright (C) 2016 OpenWrt.org
+# Copyright (C) 2016 - 2017 OpenWrt.org
# Copyright (C) 2016 Cesnet, z.s.p.o.
#
# This is free software, licensed under the GNU General Public License v2.
PKG_NAME:=chan-sccp-b
PKG_REV:=6728
PKG_VERSION:=v4.2.3-r$(PKG_REV)
-PKG_RELEASE:=1
+PKG_RELEASE:=2
PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=http://svn.code.sf.net/p/chan-sccp-b/code/branches/v4.2
PKG_INSTALL:=1
+PKG_BUILD_DIR:=$(BUILD_DIR)/$(PKG_NAME)-$(BUILD_VARIANT)/$(PKG_NAME)-$(PKG_VERSION)
+
include $(INCLUDE_DIR)/package.mk
define Package/chan-sccp-b/Default
CATEGORY:=Network
TITLE:=SCCP channel provider support
URL:=http://chan-sccp-b.sourceforge.net/
- DEPENDS:= +libltdl
+ DEPENDS:=+libltdl
+ PKG_BUILD_DEPENDS:=libiconv
endef
define Package/asterisk13-chan-sccp-b
Package/asterisk13-chan-sccp-b/description = $(Package/description/Default)
CONFIGURE_ARGS += \
- --enable-debug=no \
- --enable-optimization=no \
+ --enable-static \
+ --enable-section-relocation \
--enable-conference \
--enable-advanced-functions \
--enable-video
projects / feed / telephony.git / commitdiff