diff options
Diffstat (limited to 'contrib/asterisk-xip')
36 files changed, 6497 insertions, 0 deletions
diff --git a/contrib/asterisk-xip/Makefile b/contrib/asterisk-xip/Makefile new file mode 100644 index 0000000000..3eff7aa965 --- /dev/null +++ b/contrib/asterisk-xip/Makefile @@ -0,0 +1,2020 @@ +# +# Copyright (C) 2007 OpenWrt.org +# +# This is free software, licensed under the GNU General Public License v2. +# See /LICENSE for more information. +# +# $Id: Makefile 13712 2008-12-21 20:34:15Z zandbelt $ + +include $(TOPDIR)/rules.mk + +PKG_NAME:=asterisk +PKG_VERSION:=1.4.22 +PKG_RELEASE:=3 + +PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.gz +PKG_SOURCE_URL:=http://downloads.digium.com/pub/asterisk/releases/ +PKG_MD5SUM:=7626febc4a01e16e012dfccb9e4ab9d2 + +PKG_BUILD_DEPENDS:= libopenh323 pwlib gsm libvorbis + +include $(INCLUDE_DIR)/package.mk + +STAMP_CONFIGURED:=$(STAMP_CONFIGURED)_$(call confvar, \ + CONFIG_PACKAGE_asterisk14-xip CONFIG_PACKAGE_asterisk14-xip-mini \ + CONFIG_PACKAGE_asterisk14-xip-chan-alsa CONFIG_PACKAGE_asterisk14-xip-chan-gtalk \ + CONFIG_PACKAGE_asterisk14-xip-chan-h323 CONFIG_PACKAGE_asterisk14-xip-chan-mgcp \ + CONFIG_PACKAGE_asterisk14-xip-chan-skinny CONFIG_PACKAGE_asterisk14-xip-codec-ilbc \ + CONFIG_PACKAGE_asterisk14-xip-codec-lpc10 CONFIG_PACKAGE_asterisk14-xip-codec-speex \ + CONFIG_PACKAGE_asterisk14-xip-pbx-dundi CONFIG_PACKAGE_asterisk14-xip-res-agi \ + CONFIG_PACKAGE_asterisk14-xip-res-crypto CONFIG_PACKAGE_asterisk14-xip-pgsql \ + CONFIG_PACKAGE_asterisk14-xip-sqlite CONFIG_PACKAGE_asterisk14-xip-voicemail \ + CONFIG_PACKAGE_asterisk14-xip-sounds \ +) + +define Package/asterisk14-xip/Default + SUBMENU:=asterisk14-xip (Complete Open Source PBX), v1.4.x + SECTION:=net + CATEGORY:=Network + URL:=http://www.asterisk.org/ +endef + +define Package/asterisk14-xip/Default/description + Asterisk is a complete PBX in software. It provides all of the features + you would expect from a PBX and more. Asterisk does voice over IP in three + protocols, and can interoperate with almost all standards-based telephony + equipment using relatively inexpensive hardware. +endef + +define Package/asterisk14-xip-core +$(call Package/asterisk14-xip/Default) + TITLE:=Asterisk Core + DEPENDS:=+libncurses +libpopt +libpthread @!TARGET_avr32 +endef + +define Package/asterisk14-xip-core/description +$(call Package/asterisk14-xip/Default/description) +Asterisk Core + codec_gsm + format_gsm + pbx_config Read Configuration + res_indications Tone support + app_dial + chan_local Dial Local channel +endef + +define Package/asterisk14-xip +$(call Package/asterisk14-xip/Default) + TITLE:=Complete open source PBX + DEPENDS:= +asterisk14-xip-core +asterisk14-xip-iax +asterisk14-xip-sip +asterisk14-xip-codec-ualaw +asterisk14-xip-codec-wav +asterisk14-xip-features +asterisk14-xip-moh \ + +asterisk14-xip-app-meetme +asterisk14-xip-chan-oss +asterisk14-xip-chan-alsa +asterisk14-xip-chan-gtalk +asterisk14-xip-chan-h323 +asterisk14-xip-chan-mgcp \ + +asterisk14-xip-chan-skinny +asterisk14-xip-codec-lpc10 +asterisk14-xip-codec-speex +asterisk14-xip-pbx-dundi +asterisk14-xip-res-agi +asterisk14-xip-res-crypto \ + +asterisk14-xip-pgsql +asterisk14-xip-sqlite +asterisk14-xip-voicemail +asterisk14-xip-sounds +asterisk14-xip-rawplayer +asterisk14-xip-agents +asterisk14-xip-iax \ + +asterisk14-xip-sip +asterisk14-xip-codec-wav +asterisk14-xip-codec-ualaw +asterisk14-xip-format-misc +asterisk14-xip-format-licensed +asterisk14-xip-codec-g726 \ + +asterisk14-xip-format-video +asterisk14-xip-variables +asterisk14-xip-enum +asterisk14-xip-basic +asterisk14-xip-encode +asterisk14-xip-realtime \ + +asterisk14-xip-ael +asterisk14-xip-adsi +asterisk14-xip-features +asterisk14-xip-moh +asterisk14-xip-smdi +asterisk14-xip-sounds-tt \ + +asterisk14-xip-sounds-demo +asterisk14-xip-linejack +asterisk14-xip-app-misc +asterisk14-xip-image +asterisk14-xip-sms +asterisk14-xip-icecast \ + +asterisk14-xip-mp3 +asterisk14-xip-cli +asterisk14-xip-isdn +asterisk14-xip-deprecated +asterisk14-xip-groups +asterisk14-xip-language +asterisk14-xip-spool \ + +asterisk14-xip-nbs +asterisk14-xip-alarmreceiver +asterisk14-xip-cdr +asterisk14-xip-channel +asterisk14-xip-debug +asterisk14-xip-menu-misc \ + +asterisk14-xip-festival +asterisk14-xip-send-app +asterisk14-xip-followme +asterisk14-xip-queues +asterisk14-xip-record +asterisk14-xip-privacy \ + +asterisk14-xip-ivr-util +asterisk14-xip-callerid +asterisk14-xip-speech +asterisk14-xip-detect +asterisk14-xip-controlflow @!TARGET_avr32 +endef + +define Package/asterisk14-xip/description +$(call Package/asterisk14-xip/Default/description) +endef + + +define Package/asterisk14-xip-mini +$(call Package/asterisk14-xip/Default) + TITLE:=Minimal open source PBX + DEPENDS:=+libncurses +libpthread +asterisk14-xip-core +asterisk14-xip-iax +asterisk14-xip-sip +asterisk14-xip-codec-ualaw +asterisk14-xip-codec-wav +asterisk14-xip-features +asterisk14-xip-moh +libgsm @!TARGET_avr32 +endef + +define Package/asterisk14-xip-mini/description +$(call Package/asterisk14-xip/Default/description) + This package contains only the following modules: + - app_dial + - chan_iax2 + - chan_local + - chan_sip + - codec_gsm + - codec_ulaw + - format_gsm + - format_pcm + - format_wav + - format_wav_gsm + - pbx_config + - res_features + - res_musiconhold +endef + + +define Package/asterisk14-xip-app-meetme +$(call Package/asterisk14-xip/Default) + TITLE:=conferencing support + DEPENDS:= +asterisk14-xip-core +zaptel14-libtonezone +endef + +define Package/asterisk14-xip-app-meetme/description +$(call Package/asterisk14-xip/Default/description) + This package provides the MeetMe application driver Conferencing support to + Asterisk. + app_meetme + app_page Paging multiple extensions. +endef + + +define Package/asterisk14-xip-chan-oss +$(call Package/asterisk14-xip/Default) + TITLE:=OSS soundcards support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-chan-oss/description +$(call Package/asterisk14-xip/Default/description) + This package provides the channel driver for OSS sound cards support to + Asterisk. +endef + +define Package/asterisk14-xip-chan-alsa +$(call Package/asterisk14-xip/Default) + TITLE:=ALSA soundcards support + DEPENDS:= +asterisk14-xip-core +alsa-lib +endef + +define Package/asterisk14-xip-chan-alsa/description +$(call Package/asterisk14-xip/Default/description) + This package provides the channel driver for ALSA sound cards support to + Asterisk. +endef + + +define Package/asterisk14-xip-chan-gtalk +$(call Package/asterisk14-xip/Default) + TITLE:=GTalk support + DEPENDS:= +asterisk14-xip-core +libiksemel +endef + +define Package/asterisk14-xip-chan-gtalk/description +$(call Package/asterisk14-xip/Default/description) + This package provides the channel chan_gtalk and res_jabber for GTalk + support to Asterisk. +endef + + +define Package/asterisk14-xip-chan-h323 +$(call Package/asterisk14-xip/Default) + TITLE:=H.323 support for Asterisk + DEPENDS:= +asterisk14-xip-core +uclibcxx +endef + +define Package/asterisk14-xip-chan-h323/description +$(call Package/asterisk14-xip/Default/description) + This package provides H.323 support to Asterisk. +endef + + +define Package/asterisk14-xip-chan-mgcp +$(call Package/asterisk14-xip/Default) + TITLE:=MGCP support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-chan-mgcp/description +$(call Package/asterisk14-xip/Default/description) + This package provides MGCP (Media Gateway Control Protocol) support \\\ + to Asterisk. +endef + + +define Package/asterisk14-xip-chan-skinny +$(call Package/asterisk14-xip/Default) + TITLE:=Skinny Client Control Protocol support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-chan-skinny/description +$(call Package/asterisk14-xip/Default/description) + This package provided Skinny Client Control Protocol support to \\\ + Asterisk. +endef + + +#define Package/asterisk14-xip-codec-ilbc +#$(call Package/asterisk14-xip/Default) +# TITLE:=ILBC Translator +# DEPENDS:= +asterisk14-xip-core +#endef + +#define Package/asterisk14-xip-codec-ilbc/description +#$(call Package/asterisk14-xip/Default/description) +# This package contains the ILBC (Internet Low Bitrate Codec) translator +# for Asterisk. +#endef + + +define Package/asterisk14-xip-codec-lpc10 +$(call Package/asterisk14-xip/Default) + TITLE:=LPC10 2.4kbps voice codec Translator + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-chan-lpc10/description +$(call Package/asterisk14-xip/Default/description) + This package contains the LPC10 (Linear Predictor Code) 2.4kbps voice + codec translator for Asterisk. +endef + + +define Package/asterisk14-xip-codec-speex +$(call Package/asterisk14-xip/Default) + TITLE:=Speex/PCM16 Codec Translator + DEPENDS:= +asterisk14-xip-core +libspeex +libspeexdsp +endef + +define Package/asterisk14-xip-chan-speex/description +$(call Package/asterisk14-xip/Default/description) + This package contains the Speex speech compression codec translator for + Asterisk. +endef + + +define Package/asterisk14-xip-pbx-dundi +$(call Package/asterisk14-xip/Default) + TITLE:=DUNDi support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-pbx-dundi/description +$(call Package/asterisk14-xip/Default/description) + This package provides DUNDi (Distributed Universal Number Discovery) + support to Asterisk. +endef + + +define Package/asterisk14-xip-res-agi +$(call Package/asterisk14-xip/Default) + TITLE:=AGI support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-res-agi/description +$(call Package/asterisk14-xip/Default/description) + This package provides AGI (Asterisk Gateway Interface) support to + Asterisk. +endef + + +define Package/asterisk14-xip-res-crypto +$(call Package/asterisk14-xip/Default) + TITLE:=Cryptographic Digital Signatures support + DEPENDS:= +asterisk14-xip-core +libopenssl +endef + +define Package/asterisk14-xip-res-crypto/description +$(call Package/asterisk14-xip/Default/description) + This package provides Cryptographic Digital Signatures support to + Asterisk. +endef + + +define Package/asterisk14-xip-pgsql +$(call Package/asterisk14-xip/Default) + TITLE:=PostgreSQL support + DEPENDS:= +asterisk14-xip-core +libpq +endef + +define Package/asterisk14-xip-pgsql/description +$(call Package/asterisk14-xip/Default/description) + This package contains PostgreSQL support modules for Asterisk. +endef + + +define Package/asterisk14-xip-sqlite +$(call Package/asterisk14-xip/Default) + TITLE:=SQLite modules + DEPENDS:= +asterisk14-xip-core +libsqlite2 +endef + +define Package/asterisk14-xip-sqlite/description +$(call Package/asterisk14-xip/Default/description) + This package contains SQLite support modules for Asterisk. +endef + + +define Package/asterisk14-xip-sounds +$(call Package/asterisk14-xip/Default) + TITLE:=Sound files + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-sounds/description +$(call Package/asterisk14-xip/Default/description) + This package contains sound files for Asterisk. +endef + + +define Package/asterisk14-xip-voicemail +$(call Package/asterisk14-xip/Default) + TITLE:=Voicemail support + DEPENDS:= +asterisk14-xip-core +asterisk14-xip-adsi +endef + +define Package/asterisk14-xip-voicemail/description +$(call Package/asterisk14-xip/Default/description) + This package contains voicemail related modules for Asterisk. +endef + +define Package/asterisk14-xip-rawplayer +$(call Package/asterisk14-xip/Default) + TITLE:=Play raw files for asterisk +endef + +define Package/asterisk14-xip-rawplayer/description + Contains the rawplayer utility for asterisk +endef + +define Package/asterisk14-xip-agents +$(call Package/asterisk14-xip/Default) + TITLE:=Support for user Agents + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-agents/description +$(call Package/asterisk14-xip/Default/description) +Support for user Agents + chan_agent +endef + +define Package/asterisk14-xip-iax +$(call Package/asterisk14-xip/Default) + TITLE:=IAX2 Channel support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-iax/description +$(call Package/asterisk14-xip/Default/description) +IAX2 Channel support + chan_iax2 +endef + +define Package/asterisk14-xip-sip +$(call Package/asterisk14-xip/Default) + TITLE:=SIP Channel support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-sip/description +$(call Package/asterisk14-xip/Default/description) +SIP Channel support + chan_sip +endef + +define Package/asterisk14-xip-codec-wav +$(call Package/asterisk14-xip/Default) + TITLE:=WAV/PCM Codecs + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-codec-wav/description +$(call Package/asterisk14-xip/Default/description) +WAV/PCM Codecs + codec_adpcm + format_pcm + format_wav_gsm Microsoft Proprietary Wave GSM format + format_wav +endef + +define Package/asterisk14-xip-codec-ualaw +$(call Package/asterisk14-xip/Default) + TITLE:=Ulaw/Alaw Codec support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-codec-ualaw/description +$(call Package/asterisk14-xip/Default/description) +Ulaw/Alaw Codec support + codec_alaw + codec_a_mu A-Law and MUlaw direct coder/Decoder + codec_ulaw +endef + +define Package/asterisk14-xip-format-misc +$(call Package/asterisk14-xip/Default) + TITLE:=Misc pass-through formats + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-format-misc/description +$(call Package/asterisk14-xip/Default/description) +Misc pass-through formats + format_sln + format_vox + format_ilbc iLBC +endef + +define Package/asterisk14-xip-format-licensed +$(call Package/asterisk14-xip/Default) + TITLE:=Licenses and Patented Formats Passthrough + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-format-licensed/description +$(call Package/asterisk14-xip/Default/description) +Licenses and Patented Formats Passthrough + format_g726 + format_g723 + format_g729 +endef + +define Package/asterisk14-xip-codec-g726 +$(call Package/asterisk14-xip/Default) + TITLE:=G726 Codec (requires license) + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-codec-g726/description +$(call Package/asterisk14-xip/Default/description) +G726 Codec (requires license) + codec_g726 +endef + +define Package/asterisk14-xip-format-video +$(call Package/asterisk14-xip/Default) + TITLE:=Video formats + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-format-video/description +$(call Package/asterisk14-xip/Default/description) +Video formats + format_h263 + format_h264 +endef + +define Package/asterisk14-xip-variables +$(call Package/asterisk14-xip/Default) + TITLE:=Read Variables and environment + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-variables/description +$(call Package/asterisk14-xip/Default/description) +Read Variables and environment + func_db + func_global + func_env + func_timeout Control timeout values +endef + +define Package/asterisk14-xip-enum +$(call Package/asterisk14-xip/Default) + TITLE:=DNS Enum support to find alternate call route + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-enum/description +$(call Package/asterisk14-xip/Default/description) +DNS Enum support to find alternate call route + func_enum Use DNS to find alternate calling method +endef + +define Package/asterisk14-xip-basic +$(call Package/asterisk14-xip/Default) + TITLE:=Basic functions + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-basic/description +$(call Package/asterisk14-xip/Default/description) +Basic functions + func_logic + func_math + func_strings + func_rand + func_cut +endef + +define Package/asterisk14-xip-encode +$(call Package/asterisk14-xip/Default) + TITLE:=Support for string encoding/hashing + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-encode/description +$(call Package/asterisk14-xip/Default/description) +Support for string encoding/hashing + func_base64 + func_md5 + func_sha1 + func_uri +endef + +define Package/asterisk14-xip-realtime +$(call Package/asterisk14-xip/Default) + TITLE:=Asterisk Realtime support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-realtime/description +$(call Package/asterisk14-xip/Default/description) +Asterisk Realtime support + func_realtime + pbx_realtime + app_realtime 'Realtime' support +endef + +define Package/asterisk14-xip-ael +$(call Package/asterisk14-xip/Default) + TITLE:=AEL - Asterisk Extension Language compiler support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-ael/description +$(call Package/asterisk14-xip/Default/description) +AEL - Asterisk Extension Language compiler support + pbx_ael Asterisk Extension Language compiler +endef + +define Package/asterisk14-xip-adsi +$(call Package/asterisk14-xip/Default) + TITLE:=ADSI Support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-adsi/description +$(call Package/asterisk14-xip/Default/description) +ADSI Support + res_adsi + app_adsiprog +endef + +define Package/asterisk14-xip-features +$(call Package/asterisk14-xip/Default) + TITLE:=Call Features / Parking + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-features/description +$(call Package/asterisk14-xip/Default/description) +Call Features / Parking + res_features Features support. + app_transfer + app_parkandannounce + res_monitor Record channels +endef + +define Package/asterisk14-xip-moh +$(call Package/asterisk14-xip/Default) + TITLE:=Music On Hold support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-moh/description +$(call Package/asterisk14-xip/Default/description) +Music On Hold support + res_musiconhold + func_moh +endef + +define Package/asterisk14-xip-smdi +$(call Package/asterisk14-xip/Default) + TITLE:=Simple Message Desk Interface + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-smdi/description +$(call Package/asterisk14-xip/Default/description) +Simple Message Desk Interface + res_smdi Simple Message Desk Interface +endef + +define Package/asterisk14-xip-sounds-tt +$(call Package/asterisk14-xip/Default) + TITLE:=Telemarketer Torture Sounds + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-sounds-tt/description +$(call Package/asterisk14-xip/Default/description) +Telemarketer Torture Sounds +endef + +define Package/asterisk14-xip-sounds-demo +$(call Package/asterisk14-xip/Default) + TITLE:=Demo Sounds + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-sounds-demo/description +$(call Package/asterisk14-xip/Default/description) +Demo Sounds +endef + +define Package/asterisk14-xip-linejack +$(call Package/asterisk14-xip/Default) + TITLE:=M chan_phone (32,988) Linejack Cards + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-linejack/description +$(call Package/asterisk14-xip/Default/description) +M chan_phone (32,988) Linejack Cards +endef + +define Package/asterisk14-xip-app-misc +$(call Package/asterisk14-xip/Default) + TITLE:=Misc applications + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-app-misc/description +$(call Package/asterisk14-xip/Default/description) +Misc applications + app_random + app_sayunixtime + app_sendtext + app_url + app_readfile + app_system Call System application. + app_exec Exec Dialplan applications +endef + +define Package/asterisk14-xip-image +$(call Package/asterisk14-xip/Default) + TITLE:=Support for images + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-image/description +$(call Package/asterisk14-xip/Default/description) +Support for images + app_image Transmit images + format_jpeg +endef + +define Package/asterisk14-xip-sms +$(call Package/asterisk14-xip/Default) + TITLE:=SMS support + DEPENDS:= +asterisk14-xip-core +libstdcpp +endef + +define Package/asterisk14-xip-sms/description +$(call Package/asterisk14-xip/Default/description) +SMS support + app_sms +endef + +define Package/asterisk14-xip-icecast +$(call Package/asterisk14-xip/Default) + TITLE:=ICEcast support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-icecast/description +$(call Package/asterisk14-xip/Default/description) +ICEcast support + app_ices Icecast / Ices support +endef + +define Package/asterisk14-xip-mp3 +$(call Package/asterisk14-xip/Default) + TITLE:=MP3 Support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-mp3/description +$(call Package/asterisk14-xip/Default/description) +MP3 Support + app_mp3 +endef + +define Package/asterisk14-xip-cli +$(call Package/asterisk14-xip/Default) + TITLE:=CLI Apps and events + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-cli/description +$(call Package/asterisk14-xip/Default/description) +CLI Apps and events + app_userevent + res_clioriginate Originate a call on the CLI + res_convert File format conversion +endef + +define Package/asterisk14-xip-isdn +$(call Package/asterisk14-xip/Default) + TITLE:=ISDN transfer capability + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-isdn/description +$(call Package/asterisk14-xip/Default/description) +ISDN transfer capability + app_settransfercapability ISDN transfer capability +endef + +define Package/asterisk14-xip-deprecated +$(call Package/asterisk14-xip/Default) + TITLE:=Deprecated + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-deprecated/description +$(call Package/asterisk14-xip/Default/description) +Deprecated + app_db Deprecated - use func_db instead +endef + +define Package/asterisk14-xip-groups +$(call Package/asterisk14-xip/Default) + TITLE:=Group Functions + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-groups/description +$(call Package/asterisk14-xip/Default/description) +Group Functions + func_groupcount +endef + +define Package/asterisk14-xip-language +$(call Package/asterisk14-xip/Default) + TITLE:=Language support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-language/description +$(call Package/asterisk14-xip/Default/description) +Language support + func_language +endef + +define Package/asterisk14-xip-spool +$(call Package/asterisk14-xip/Default) + TITLE:=Spool Directory of Outgoing calls + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-spool/description +$(call Package/asterisk14-xip/Default/description) +Spool Directory of Outgoing calls + pbx_spool Spool Directory of Outgoing calls +endef + +define Package/asterisk14-xip-nbs +$(call Package/asterisk14-xip/Default) + TITLE:=NBS stream support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-nbs/description +$(call Package/asterisk14-xip/Default/description) +NBS stream support + app_nbscat +endef + +define Package/asterisk14-xip-alarmreceiver +$(call Package/asterisk14-xip/Default) + TITLE:=SIA Contact ID Alarm receiver + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-alarmreceiver/description +$(call Package/asterisk14-xip/Default/description) +SIA Contact ID Alarm receiver + app_alarmreceiver +endef + +define Package/asterisk14-xip-cdr +$(call Package/asterisk14-xip/Default) + TITLE:=CDR Support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-cdr/description +$(call Package/asterisk14-xip/Default/description) +CDR Support + app_cdr + app_forkcdr + app_setcdruserfield + cdr_csv + cdr_custom + cdr_manager + func_cdr +endef + +define Package/asterisk14-xip-channel +$(call Package/asterisk14-xip/Default) + TITLE:=Channel functions + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-channel/description +$(call Package/asterisk14-xip/Default/description) +Channel functions + app_chanisavail + app_channelredirect + app_chanspy + func_channel + app_softhangup + app_directed_pickup Pickup a (specific) ringing extensions +endef + +define Package/asterisk14-xip-debug +$(call Package/asterisk14-xip/Default) + TITLE:=Debugging tools + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-debug/description +$(call Package/asterisk14-xip/Default/description) +Debugging tools + app_echo + pbx_loopback + app_dumpchan Dump information about the calling channel + app_verbose + app_test AIX Server/client testing +endef + +define Package/asterisk14-xip-menu-misc +$(call Package/asterisk14-xip/Default) + TITLE:=Special menu applications + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-menu-misc/description +$(call Package/asterisk14-xip/Default/description) +Special menu applications + app_controlplayback + app_directory + app_dictate +endef + +define Package/asterisk14-xip-festival +$(call Package/asterisk14-xip/Default) + TITLE:=Festival support + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-festival/description +$(call Package/asterisk14-xip/Default/description) +Festival support + app_festival +endef + +define Package/asterisk14-xip-send-app +$(call Package/asterisk14-xip/Default) + TITLE:=Misc tone sending applications + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-send-app/description +$(call Package/asterisk14-xip/Default/description) +Misc tone sending applications + app_flash Send a flash + app_senddtmf Send dtmf + app_milliwatt + app_morsecode + app_zapateller Generate tone to block telemarketers +endef + +define Package/asterisk14-xip-followme +$(call Package/asterisk14-xip/Default) + TITLE:=Followme - Call forwarding + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-followme/description +$(call Package/asterisk14-xip/Default/description) +Followme - Call forwarding + app_followme +endef + + +define Package/asterisk14-xip-queues +$(call Package/asterisk14-xip/Default) + TITLE:=Call queues + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-queues/description +$(call Package/asterisk14-xip/Default/description) +Call queues + app_queue +endef + +define Package/asterisk14-xip-record +$(call Package/asterisk14-xip/Default) + TITLE:=Call recording + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-record/description +$(call Package/asterisk14-xip/Default/description) +Call recording + app_record + app_mixmonitor Records The audio on the current channel to the specified file. +endef + +define Package/asterisk14-xip-privacy +$(call Package/asterisk14-xip/Default) + TITLE:=Call Privacy - Prompt for unknown numbers. + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-privacy/description +$(call Package/asterisk14-xip/Default/description) +Call Privacy - Prompt for unknown numbers. + app_privacy Prompt for missing calling number +endef + +define Package/asterisk14-xip-ivr-util +$(call Package/asterisk14-xip/Default) + TITLE:=Utilities for creating IVR + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-ivr-util/description +$(call Package/asterisk14-xip/Default/description) +Utilities for creating IVR + app_read Read a DTMF response + app_authenticate Authenticate a user + app_externalivr IVR Using an External process. + app_disa Directed Inward Sysytem Access - Allow access to your internal dialplan with password +endef + + +define Package/asterisk14-xip-callerid +$(call Package/asterisk14-xip/Default) + TITLE:=Callerid related functions. + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-callerid/description +$(call Package/asterisk14-xip/Default/description) +Callerid related functions. + app_setcallerid + func_callerid + app_lookupblacklist + app_lookupcidname +endef + +define Package/asterisk14-xip-speech +$(call Package/asterisk14-xip/Default) + TITLE:=Interface to Speech recognition programs + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-speech/description +$(call Package/asterisk14-xip/Default/description) +Interface to Speech recognition programs + app_speech_utils + res_speech +endef + +define Package/asterisk14-xip-detect +$(call Package/asterisk14-xip/Default) + TITLE:=Detect coditions + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-detect/description +$(call Package/asterisk14-xip/Default/description) +Detect coditions + app_amd Answer machine detect + app_talkdetect + app_waitforring + app_waitforsilence +endef + +define Package/asterisk14-xip-controlflow +$(call Package/asterisk14-xip/Default) + TITLE:=Advanced Control Flow + DEPENDS:= +asterisk14-xip-core +endef + +define Package/asterisk14-xip-controlflow/description +$(call Package/asterisk14-xip/Default/description) +Advanced Control Flow + app_while + app_macro Dialplan Macros + app_stack Stack routines (Gosub, Return) +endef + + +CONFIGURE_ARGS+= \ + --without-curl \ + --without-curses \ + --with-gsm="$(STAGING_DIR)/usr" \ + --without-imap \ + --without-isdnnet \ + --without-kde \ + --without-misdn \ + --without-nbs \ + --with-ncurses="$(STAGING_DIR)/usr" \ + --without-netsnmp \ + --without-newt \ + --without-odbc \ + --without-ogg \ + --without-osptk \ + --with-popt="$(STAGING_DIR)/usr" \ + --without-pri \ + --without-qt \ + --without-radius \ + --without-spandsp \ + --without-suppserv \ + --without-tds \ + --without-termcap \ + --without-tinfo \ + --without-vorbis \ + --without-vpb \ + --with-z="$(STAGING_DIR)/usr" \ + +EXTRA_CFLAGS:= $(TARGET_CPPFLAGS) +EXTRA_LDFLAGS:= $(TARGET_LDFLAGS) + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-app-meetme),) + CONFIGURE_ARGS+= \ + --with-tonezone="$(STAGING_DIR)/usr" --with-zaptel="$(STAGING_DIR)/usr" +else + CONFIGURE_ARGS+= \ + --without-tonezone --without-zaptel +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-chan-alsa),) + CONFIGURE_ARGS+= \ + --with-asound="$(STAGING_DIR)/usr" +else + CONFIGURE_ARGS+= \ + --without-asound +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-chan-oss),) + CONFIGURE_ARGS+= \ + --with-oss +else + CONFIGURE_ARGS+= \ + --without-oss +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-chan-gtalk),) + CONFIGURE_ARGS+= \ + --with-gnutls="$(STAGING_DIR)/usr" \ + --with-iksemel="$(STAGING_DIR)/usr" + SITE_VARS+= \ + ac_cv_lib_iksemel_iks_start_sasl=yes \ + ac_cv_lib_gnutls_gnutls_bye=yes +else + CONFIGURE_ARGS+= \ + --without-gnutls \ + --without-iksemel +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-chan-h323),) + CONFIGURE_ARGS+= \ + --with-h323="$(BUILD_DIR)/openh323" \ + --with-pwlib="$(BUILD_DIR)/pwlib" + CONFIGURE_VARS+= \ + LIBS="$$$$LIBS -luClibc++ -ldl -lpthread" + + define Build/Compile/chan-h323 + $(MAKE) -C "$(PKG_BUILD_DIR)/channels/h323" \ + $(TARGET_CONFIGURE_OPTS) \ + CXXLIBS="-nodefaultlibs -luClibc++" \ + optnoshared + endef +else + CONFIGURE_ARGS+= \ + --without-h323 \ + --without-pwlib +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-codec-speex),) + CONFIGURE_ARGS+= \ + --with-speex="$(STAGING_DIR)/usr" + SITE_VARS+= \ + ac_cv_lib_speex_speex_encode=yes + EXTRA_CFLAGS+= -I$(STAGING_DIR)/usr/include/speex +else + CONFIGURE_ARGS+= \ + --without-speex +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-res-crypto),) + CONFIGURE_ARGS+= \ + --with-ssl="$(STAGING_DIR)/usr" +else + CONFIGURE_ARGS+= \ + --without-ssl +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-pgsql),) + CONFIGURE_ARGS+= \ + --with-postgres="$(STAGING_DIR)/usr" +else + CONFIGURE_ARGS+= \ + --without-postgres +endif + +ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk14-xip-sqlite),) + CONFIGURE_ARGS+= \ + --with-sqlite="$(STAGING_DIR)/usr" +else + CONFIGURE_ARGS+= \ + --without-sqlite +endif + + +define Build/Configure + -rm $(PKG_BUILD_DIR)/menuselect.makeopts + ( cd $(PKG_BUILD_DIR); ./bootstrap.sh ) + $(call Build/Configure/Default,,$(SITE_VARS)) +endef + +define Build/Compile + $(MAKE) -C "$(PKG_BUILD_DIR)" \ + include/asterisk/version.h \ + include/asterisk/buildopts.h defaults.h \ + makeopts.embed_rules + $(call Build/Compile/chan-h323) + ASTCFLAGS="$(EXTRA_CFLAGS) -DLOW_MEMORY $(TARGET_CFLAGS)" \ + ASTLDFLAGS="$(EXTRA_LDFLAGS)" \ + $(MAKE) -C "$(PKG_BUILD_DIR)" \ + ASTVARLIBDIR="/usr/lib/asterisk" \ + NOISY_BUILD="1" \ + DEBUG="" \ + OPTIMIZE="" \ + DESTDIR="$(PKG_INSTALL_DIR)" \ + all install samples + $(SED) 's|/var/lib/asterisk|/usr/lib/asterisk|g' $(PKG_INSTALL_DIR)/etc/asterisk/musiconhold.conf + + $(TARGET_CC) -O2 $(PKG_BUILD_DIR)/contrib/utils/rawplayer.c -o $(PKG_BUILD_DIR)/rawplayer +endef + +define Build/InstallDev + mkdir -p $(1)/usr/include/asterisk/ + $(CP) $(PKG_INSTALL_DIR)/usr/include/asterisk/*.h $(1)/usr/include/asterisk/ + $(CP) $(PKG_INSTALL_DIR)/usr/include/asterisk.h $(1)/usr/include/ +endef + +define Package/asterisk14-xip-core/conffiles +/etc/asterisk/asterisk.conf +/etc/asterisk/codecs.conf +/etc/asterisk/dnsmgr.conf +/etc/asterisk/extconfig.conf +/etc/asterisk/extensions.conf +/etc/asterisk/http.conf +/etc/asterisk/indications.conf +/etc/asterisk/logger.conf +/etc/asterisk/manager.conf +/etc/asterisk/modules.conf +/etc/asterisk/say.conf +/etc/asterisk/sla.conf +/etc/asterisk/users.conf +endef + +define Package/asterisk14-xip-core/install + $(INSTALL_DIR) $(1)/etc/asterisk + for f in users.conf extensions.conf say.conf asterisk.conf codecs.conf dnsmgr.conf extconfig.conf http.conf indications.conf logger.conf sla.conf manager.conf ; do \ + $(CP) $(PKG_INSTALL_DIR)/etc/asterisk/$$$$f $(1)/etc/asterisk/ ; \ + done + $(INSTALL_DATA) ./files/modules.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk + $(INSTALL_DIR) $(1)/usr/lib/asterisk/keys + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in codec_gsm format_gsm pbx_config res_indications app_dial chan_local ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/moh + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(INSTALL_DIR) $(1)/usr/sbin + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/sbin/asterisk $(1)/usr/sbin/ + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/sbin/safe_asterisk $(1)/usr/sbin/ + $(INSTALL_DIR) $(1)/etc/default + $(INSTALL_DATA) ./files/asterisk.default $(1)/etc/default/asterisk + $(INSTALL_DIR) $(1)/etc/init.d + $(INSTALL_BIN) ./files/asterisk.init $(1)/etc/init.d/asterisk + $(INSTALL_DIR) $(1)/usr/lib/asterisk/uci + $(CP) ./files/uci/* $1/usr/lib/asterisk/uci + $(INSTALL_DIR) $(1)/etc/asterisk/macros + $(CP) ./files/macros/* $1/etc/asterisk/macros +endef + +define Package/asterisk14-xip-core/postinst +#!/bin/sh +ROOT=`echo $${PKG_ROOT} | sed 's:[\/]:\\\&:g' -` +/bin/sed -i 's/\ \/etc/\ '$${ROOT}'etc/g' $${PKG_ROOT}/etc/asterisk/asterisk.conf +/bin/sed -i 's/\ \/var\/spool/\ '$${ROOT}'var\/spool/g' $${PKG_ROOT}/etc/asterisk/asterisk.conf +/bin/sed -i 's/\ \/var\/log/\ '$${ROOT}'var\/log/g' $${PKG_ROOT}/etc/asterisk/asterisk.conf +/bin/sed -i 's/\ \/usr/\ '$${ROOT}'usr/g' $${PKG_ROOT}/etc/asterisk/asterisk.conf +/bin/sed -i 's/^DEST=/DEST='$${ROOT}'/g' $${PKG_ROOT}/etc/init.d/asterisk +/bin/sed -i 's/OPTIONS=\"\"/OPTIONS=\"-C\ '$${ROOT}'etc\/asterisk\/asterisk.conf\"/g' $${PKG_ROOT}/etc/default/asterisk +mkdir -p $${PKG_ROOT}/etc/asterisk/conf.d +cd $${PKG_ROOT}/etc/asterisk/conf.d +ln -s ../../../usr/lib/asterisk/uci/voicemailconf 10-voicemail +ln -s ../../../usr/lib/asterisk/uci/mohconf 15-moh +ln -s ../../../usr/lib/asterisk/uci/featureconf 20-features +ln -s ../../../usr/lib/asterisk/uci/lastcall 25-lastcall +ln -s ../../../usr/lib/asterisk/uci/meetmeconf 30-meetme +ln -s ../../../usr/lib/asterisk/uci/sipiaxconf 35-sipiax +ln -s ../../../usr/lib/asterisk/uci/talkclock 40-talkclock +endef + +define Package/asterisk14-xip/install +endef + +define Package/asterisk14-xip-mini/install +endef + +define Package/asterisk14-xip-app-meetme/conffiles +/etc/asterisk/meetme.conf +endef + +define Package/asterisk14-xip-app-meetme/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/meetme.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_meetme app_page ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/conf-* $(1)/usr/lib/asterisk/sounds/ +endef + + +define Package/asterisk14-xip-chan-oss/conffiles +/etc/asterisk/oss.conf +endef + +define Package/asterisk14-xip-chan-oss/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/oss.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_oss.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-app-meetme/conffiles +/etc/asterisk/meetme.conf +endef + +define Package/asterisk14-xip-app-meetme/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/meetme.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_meetme.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-oss/conffiles +/etc/asterisk/oss.conf +endef + +define Package/asterisk14-xip-chan-oss/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/oss.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_oss.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-alsa/conffiles +/etc/asterisk/alsa.conf +endef + +define Package/asterisk14-xip-chan-alsa/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/alsa.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_alsa.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-gtalk/conffiles +/etc/asterisk/gtalk.conf +/etc/asterisk/jabber.conf +endef + +define Package/asterisk14-xip-chan-gtalk/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/gtalk.conf $(1)/etc/asterisk/ + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/jabber.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_gtalk.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/res_jabber.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-h323/conffiles +/etc/asterisk/h323.conf +endef + +define Package/asterisk14-xip-chan-h323/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/h323.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_h323.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-mgcp/install +/etc/asterisk/mgcp.conf +endef + +define Package/asterisk14-xip-chan-mgcp/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/mgcp.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_mgcp.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-chan-skinny/conffiles +/etc/asterisk/skinny.conf +endef + +define Package/asterisk14-xip-chan-skinny/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/skinny.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_skinny.so $(1)/usr/lib/asterisk/modules/ +endef + + +#define Package/asterisk14-xip-codec-ilbc/install +# $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules +# $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/codec_ilbc.so $(1)/usr/lib/asterisk/modules/ +# $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/format_ilbc.so $(1)/usr/lib/asterisk/modules/ +#endef + + +define Package/asterisk14-xip-codec-lpc10/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/codec_lpc10.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-codec-speex/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/codec_speex.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-pbx-dundi/conffiles +/etc/asterisk/dundi.conf +endef + +define Package/asterisk14-xip-pbx-dundi/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/dundi.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/pbx_dundi.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-res-agi/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/agi-bin + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/res_agi.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-res-crypto/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/res_crypto.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-pgsql/conffiles +/etc/asterisk/cdr_pgsql.conf +/etc/asterisk/res_pgsql.conf +endef + +define Package/asterisk14-xip-pgsql/install + $(INSTALL_DIR) $(1)/etc/asterisk + install -m0600 $(PKG_INSTALL_DIR)/etc/asterisk/cdr_pgsql.conf $(1)/etc/asterisk/ + install -m0600 $(PKG_INSTALL_DIR)/etc/asterisk/res_pgsql.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/cdr_pgsql.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/res_config_pgsql.so $(1)/usr/lib/asterisk/modules/ +endef + + +define Package/asterisk14-xip-sqlite/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/cdr_sqlite.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-sounds/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/* $(1)/usr/lib/asterisk/sounds/ + rm -f $(1)/usr/lib/asterisk/sounds/vm-* + rm -f $(1)/usr/lib/asterisk/sounds/x + rm -f $(1)/usr/lib/asterisk/sounds/dir-* + rm -f $(1)/usr/lib/asterisk/sounds/dictate/* + rm -f $(1)/usr/lib/asterisk/sounds/followme/* + rm -f $(1)/usr/lib/asterisk/sounds/conf-* + rm -f $(1)/usr/lib/asterisk/sounds/queue-* + rm -f $(1)/usr/lib/asterisk/sounds/priv* + rm -f $(1)/usr/lib/asterisk/sounds/auth-* + rm -f $(1)/usr/lib/asterisk/sounds/agent-* + rm -f $(1)/usr/lib/asterisk/sounds/tt-* + rm -f $(1)/usr/lib/asterisk/sounds/demo-* +endef + +define Package/asterisk14-xip-voicemail/conffiles +/etc/asterisk/voicemail.conf +endef + +define Package/asterisk14-xip-voicemail/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/voicemail.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/*voicemail.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/vm-*.gsm $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-rawplayer/install + $(INSTALL_DIR) $(1)/usr/bin + $(INSTALL_BIN) $(PKG_BUILD_DIR)/rawplayer \ + $(1)/usr/bin +endef + +define Package/asterisk14-xip-agents/conffiles +/etc/asterisk/agents.conf +endef + +define Package/asterisk14-xip-agents/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/agents.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_agent.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/agent-* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-iax/conffiles +/etc/asterisk/iax.conf +/etc/asterisk/iaxprov.conf +endef + +define Package/asterisk14-xip-iax/install + $(INSTALL_DIR) $(1)/etc/asterisk + for f in iax.conf iaxprov.conf ; do \ + $(CP) $(PKG_INSTALL_DIR)/etc/asterisk/$$$$f $(1)/etc/asterisk/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_iax2.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-sip/conffiles +/etc/asterisk/sip.conf +/etc/asterisk/sip_notify.conf +/etc/asterisk/rtp.conf +/etc/asterisk/udptl.conf +endef + +define Package/asterisk14-xip-sip/install + $(INSTALL_DIR) $(1)/etc/asterisk + for f in sip.conf sip_notify.conf rtp.conf udptl.conf ; do \ + $(CP) $(PKG_INSTALL_DIR)/etc/asterisk/$$$$f $(1)/etc/asterisk/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/chan_sip.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-codec-wav/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in codec_adpcm format_pcm format_wav_gsm format_wav ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-codec-ualaw/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in codec_alaw codec_a_mu codec_ulaw ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-format-misc/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in format_sln format_vox format_ilbc ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-format-licensed/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in format_g726 format_g723 format_g729 ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-codec-g726/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/codec_g726.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-format-video/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in format_h263 format_h264 ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-variables/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in func_db func_global func_env func_timeout ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-enum/conffiles +/etc/asterisk/enum.conf +endef + +define Package/asterisk14-xip-enum/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/enum.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/func_enum.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-basic/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in func_logic func_math func_strings func_rand func_cut ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-encode/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in func_base64 func_md5 func_sha1 func_uri ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-realtime/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in func_realtime pbx_realtime app_realtime ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-ael/conffiles +/etc/asterisk/extensions.ael +endef + +define Package/asterisk14-xip-ael/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/extensions.ael $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/pbx_ael.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/sbin + $(CP) $(PKG_INSTALL_DIR)/usr/sbin/aelparse $(1)/usr/sbin/ +endef + +define Package/asterisk14-xip-adsi/conffiles +/etc/asterisk/adsi.conf +endef + +define Package/asterisk14-xip-adsi/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/adsi.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in res_adsi app_adsiprog ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-features/conffiles +/etc/asterisk/features.conf +endef + +define Package/asterisk14-xip-features/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/features.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in res_features app_transfer app_parkandannounce res_monitor ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-moh/conffiles +/etc/asterisk/musiconhold.conf +endef + +define Package/asterisk14-xip-moh/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/musiconhold.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in res_musiconhold func_moh ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/sbin + $(CP) $(PKG_INSTALL_DIR)/usr/sbin/streamplayer $(1)/usr/sbin/ +endef + +define Package/asterisk14-xip-smdi/conffiles +/etc/asterisk/smdi.conf +endef + +define Package/asterisk14-xip-smdi/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/smdi.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/res_smdi.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-sounds-tt/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/tt-* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-sounds-demo/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/demo-* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-linejack/conffiles +/etc/asterisk/phone.conf +endef + +define Package/asterisk14-xip-linejack/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/phone.conf $(1)/etc/asterisk/ +endef + +define Package/asterisk14-xip-app-misc/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_random app_sayunixtime app_sendtext app_url app_readfile app_system app_exec ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-image/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_image format_jpeg ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-sms/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_sms.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/sbin + $(CP) $(PKG_INSTALL_DIR)/usr/sbin/smsq $(1)/usr/sbin/ +endef + +define Package/asterisk14-xip-icecast/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_ices.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-mp3/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_mp3.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-cli/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_userevent res_clioriginate res_convert ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-isdn/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_settransfercapability.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-deprecated/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_db.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-groups/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/func_groupcount.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-language/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/func_language.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-spool/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/pbx_spool.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-nbs/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_nbscat.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-alarmreceiver/conffiles +/etc/asterisk/alarmreceiver.conf +endef + +define Package/asterisk14-xip-alarmreceiver/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/alarmreceiver.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_alarmreceiver.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-cdr/conffiles +/etc/asterisk/cdr.conf +/etc/asterisk/cdr_custom.conf +/etc/asterisk/cdr_manager.conf +endef + +define Package/asterisk14-xip-cdr/install + $(INSTALL_DIR) $(1)/etc/asterisk + for f in cdr.conf cdr_custom.conf cdr_manager.conf ; do \ + $(CP) $(PKG_INSTALL_DIR)/etc/asterisk/$$$$f $(1)/etc/asterisk/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_cdr app_forkcdr app_setcdruserfield cdr_csv cdr_custom cdr_manager func_cdr ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-channel/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_chanisavail app_channelredirect app_chanspy func_channel app_softhangup app_directed_pickup ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-debug/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_echo pbx_loopback app_dumpchan app_verbose app_test ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-menu-misc/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_controlplayback app_directory app_dictate ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + for f in dir-* dictate/* ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/$$$$f $(1)/usr/lib/asterisk/sounds/ ; \ + done +endef + +define Package/asterisk14-xip-festival/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_festival.so $(1)/usr/lib/asterisk/modules/ +endef + +define Package/asterisk14-xip-send-app/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_flash app_senddtmf app_milliwatt app_morsecode app_zapateller ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-followme/conffiles +/etc/asterisk/followme.conf +endef + +define Package/asterisk14-xip-followme/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/followme.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_followme.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/followme/* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-queues/conffiles +/etc/asterisk/queues.conf +endef + +define Package/asterisk14-xip-queues/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/queues.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_queue.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/queue-* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-record/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_record app_mixmonitor ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-privacy/conffiles +/etc/asterisk/privacy.conf +endef + +define Package/asterisk14-xip-privacy/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/privacy.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + $(INSTALL_BIN) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/app_privacy.so $(1)/usr/lib/asterisk/modules/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/priv* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-ivr-util/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_read app_authenticate app_externalivr app_disa ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/auth-* $(1)/usr/lib/asterisk/sounds/ +endef + +define Package/asterisk14-xip-callerid/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_setcallerid func_callerid app_lookupblacklist app_lookupcidname ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-speech/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_speech_utils res_speech ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-detect/conffiles +/etc/asterisk/amd.conf +endef + +define Package/asterisk14-xip-detect/install + $(INSTALL_DIR) $(1)/etc/asterisk + $(INSTALL_DATA) $(PKG_INSTALL_DIR)/etc/asterisk/amd.conf $(1)/etc/asterisk/ + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_amd app_talkdetect app_waitforring app_waitforsilence ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-controlflow/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in app_while app_macro app_stack ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done +endef + +define Package/asterisk14-xip-zaptel/install + $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules + for f in chan_zap app_zapbarge app_zapscan codec_zap app_getcpeid app_zapras ; do \ + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/modules/$$$$f.so $(1)/usr/lib/asterisk/modules/ ; \ + done + $(INSTALL_DIR) $(1)/usr/lib/asterisk + $(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/firmware $(1)/usr/lib/asterisk/ +endef + +$(eval $(call BuildPackage,asterisk14-xip-core)) +$(eval $(call BuildPackage,asterisk14-xip)) +$(eval $(call BuildPackage,asterisk14-xip-mini)) +$(eval $(call BuildPackage,asterisk14-xip-app-meetme)) +$(eval $(call BuildPackage,asterisk14-xip-chan-oss)) +$(eval $(call BuildPackage,asterisk14-xip-chan-alsa)) +$(eval $(call BuildPackage,asterisk14-xip-chan-gtalk)) +$(eval $(call BuildPackage,asterisk14-xip-chan-h323)) +$(eval $(call BuildPackage,asterisk14-xip-chan-mgcp)) +$(eval $(call BuildPackage,asterisk14-xip-chan-skinny)) +#$(eval $(call BuildPackage,asterisk14-xip-codec-ilbc)) +$(eval $(call BuildPackage,asterisk14-xip-codec-lpc10)) +$(eval $(call BuildPackage,asterisk14-xip-codec-speex)) +$(eval $(call BuildPackage,asterisk14-xip-pbx-dundi)) +$(eval $(call BuildPackage,asterisk14-xip-res-agi)) +$(eval $(call BuildPackage,asterisk14-xip-res-crypto)) +$(eval $(call BuildPackage,asterisk14-xip-pgsql)) +$(eval $(call BuildPackage,asterisk14-xip-sqlite)) +$(eval $(call BuildPackage,asterisk14-xip-voicemail)) +$(eval $(call BuildPackage,asterisk14-xip-sounds)) +$(eval $(call BuildPackage,asterisk14-xip-rawplayer)) +$(eval $(call BuildPackage,asterisk14-xip-agents)) +$(eval $(call BuildPackage,asterisk14-xip-iax)) +$(eval $(call BuildPackage,asterisk14-xip-sip)) +$(eval $(call BuildPackage,asterisk14-xip-codec-wav)) +$(eval $(call BuildPackage,asterisk14-xip-codec-ualaw)) +$(eval $(call BuildPackage,asterisk14-xip-format-misc)) +$(eval $(call BuildPackage,asterisk14-xip-format-licensed)) +$(eval $(call BuildPackage,asterisk14-xip-codec-g726)) +$(eval $(call BuildPackage,asterisk14-xip-format-video)) +$(eval $(call BuildPackage,asterisk14-xip-variables)) +$(eval $(call BuildPackage,asterisk14-xip-enum)) +$(eval $(call BuildPackage,asterisk14-xip-basic)) +$(eval $(call BuildPackage,asterisk14-xip-encode)) +$(eval $(call BuildPackage,asterisk14-xip-realtime)) +$(eval $(call BuildPackage,asterisk14-xip-ael)) +$(eval $(call BuildPackage,asterisk14-xip-adsi)) +$(eval $(call BuildPackage,asterisk14-xip-features)) +$(eval $(call BuildPackage,asterisk14-xip-moh)) +$(eval $(call BuildPackage,asterisk14-xip-smdi)) +$(eval $(call BuildPackage,asterisk14-xip-sounds-tt)) +$(eval $(call BuildPackage,asterisk14-xip-sounds-demo)) +$(eval $(call BuildPackage,asterisk14-xip-linejack)) +$(eval $(call BuildPackage,asterisk14-xip-app-misc)) +$(eval $(call BuildPackage,asterisk14-xip-image)) +$(eval $(call BuildPackage,asterisk14-xip-sms)) +$(eval $(call BuildPackage,asterisk14-xip-icecast)) +$(eval $(call BuildPackage,asterisk14-xip-mp3)) +$(eval $(call BuildPackage,asterisk14-xip-cli)) +$(eval $(call BuildPackage,asterisk14-xip-isdn)) +$(eval $(call BuildPackage,asterisk14-xip-deprecated)) +$(eval $(call BuildPackage,asterisk14-xip-groups)) +$(eval $(call BuildPackage,asterisk14-xip-language)) +$(eval $(call BuildPackage,asterisk14-xip-spool)) +$(eval $(call BuildPackage,asterisk14-xip-nbs)) +$(eval $(call BuildPackage,asterisk14-xip-alarmreceiver)) +$(eval $(call BuildPackage,asterisk14-xip-cdr)) +$(eval $(call BuildPackage,asterisk14-xip-channel)) +$(eval $(call BuildPackage,asterisk14-xip-debug)) +$(eval $(call BuildPackage,asterisk14-xip-menu-misc)) +$(eval $(call BuildPackage,asterisk14-xip-festival)) +$(eval $(call BuildPackage,asterisk14-xip-send-app)) +$(eval $(call BuildPackage,asterisk14-xip-followme)) +$(eval $(call BuildPackage,asterisk14-xip-queues)) +$(eval $(call BuildPackage,asterisk14-xip-record)) +$(eval $(call BuildPackage,asterisk14-xip-privacy)) +$(eval $(call BuildPackage,asterisk14-xip-ivr-util)) +$(eval $(call BuildPackage,asterisk14-xip-callerid)) +$(eval $(call BuildPackage,asterisk14-xip-speech)) +$(eval $(call BuildPackage,asterisk14-xip-detect)) +$(eval $(call BuildPackage,asterisk14-xip-controlflow)) + + +#asterisk14-xip-core=codec_gsm format_gsm pbx_config res_indications app_dial chan_local +#asterisk14-xip-agents=chan_agent +#asterisk14-xip-iax=chan_iax2 +#asterisk14-xip-sip=chan_sip +#asterisk14-xip-codec-wav=codec_adpcm format_pcm format_wav_gsm format_wav +#asterisk14-xip-codec-ualaw=codec_alaw codec_a_mu codec_ulaw +#asterisk14-xip-format-misc=format_sln format_vox format_ilbc +#asterisk14-xip-format-licensed=format_g726 format_g723 format_g729 +#asterisk14-xip-codec-g726=codec_g726 +#asterisk14-xip-format-video=format_h263 format_h264 +#asterisk14-xip-variables=func_db func_global func_env func_timeout +#asterisk14-xip-enum=func_enum +#asterisk14-xip-pbx-dundi=pbx_dundi +#asterisk14-xip-basic=func_logic func_math func_strings func_rand func_cut +#asterisk14-xip-encode=func_base64 func_md5 func_sha1 func_uri +#asterisk14-xip-realtime=func_realtime pbx_realtime app_realtime +#asterisk14-xip-ael=pbx_ael +#asterisk14-xip-adsi=res_adsi app_adsiprog +#asterisk14-xip-features=res_features app_transfer app_parkandannounce res_monitor +#asterisk14-xip-moh=res_musiconhold func_moh +#asterisk14-xip-smdi=res_smdi +#asterisk14-xip-app-misc=app_random app_sayunixtime app_sendtext app_url app_readfile app_system app_exec +#asterisk14-xip-image=app_image format_jpeg +#asterisk14-xip-sms=app_sms +#asterisk14-xip-icecast=app_ices +#asterisk14-xip-mp3=app_mp3 +#asterisk14-xip-cli=app_userevent res_clioriginate res_convert +#asterisk14-xip-isdn=app_settransfercapability +#asterisk14-xip-deprecated=app_db +#asterisk14-xip-groups=func_groupcount +#asterisk14-xip-language=func_language +#asterisk14-xip-spool=pbx_spool +#asterisk14-xip-nbs=app_nbscat +#asterisk14-xip-alarmreceiver=app_alarmreceiver +#asterisk14-xip-cdr=app_cdr app_forkcdr app_setcdruserfield cdr_csv cdr_custom cdr_manager func_cdr +#asterisk14-xip-channel=app_chanisavail app_channelredirect app_chanspy func_channel app_softhangup app_directed_pickup +#asterisk14-xip-debug=app_echo pbx_loopback app_dumpchan app_verbose app_test +#asterisk14-xip-menu-misc=app_controlplayback app_directory app_dictate +#asterisk14-xip-festival=app_festival +#asterisk14-xip-send-app=app_flash app_senddtmf app_milliwatt app_morsecode app_zapateller +#asterisk14-xip-followme=app_followme +#asterisk14-xip-app-meetme=app_meetme app_page +#asterisk14-xip-queues=app_queue +#asterisk14-xip-record=app_record app_mixmonitor +#asterisk14-xip-privacy=app_privacy +#asterisk14-xip-ivr-util=app_read app_authenticate app_externalivr app_disa +#asterisk14-xip-callerid=app_setcallerid func_callerid app_lookupblacklist app_lookupcidname +#asterisk14-xip-speech=app_speech_utils res_speech +#asterisk14-xip-detect=app_amd app_talkdetect app_waitforring app_waitforsilence +#asterisk14-xip-controlflow=app_while app_macro app_stack +#asterisk14-xip-zaptel=chan_zap app_zapbarge app_zapscan codec_zap app_getcpeid app_zapras +#asterisk14-xip-chan-oss=chan_oss +#asterisk14-xip-chan-alsa=chan_alsa +#asterisk14-xip-chan-gtalk=chan_gtalk res_jabber +#asterisk14-xip-chan-h323=chan_h323 +#asterisk14-xip-chan-mgcp=chan_mgcp +#asterisk14-xip-chan-skinny=chan_skinny +#asterisk14-xip-chan-lpc10=chan_lpc10 +#asterisk14-xip-codec-speex=codec_speex +#asterisk14-xip-res-agi=res_agi +#asterisk14-xip-res-crypto=res_crypto +#asterisk14-xip-pgsql=cdr_pgsql res_config_pgsql +#asterisk14-xip-sqlite=cdr_sqlite +#asterisk14-xip-voicemail=app_hasnewvoicemail app_voicemail diff --git a/contrib/asterisk-xip/files/asterisk.default b/contrib/asterisk-xip/files/asterisk.default new file mode 100644 index 0000000000..9d046c42da --- /dev/null +++ b/contrib/asterisk-xip/files/asterisk.default @@ -0,0 +1,4 @@ +## startup options for /etc/init.d/asterisk + +ENABLE_ASTERISK="yes" +OPTIONS="" diff --git a/contrib/asterisk-xip/files/asterisk.init b/contrib/asterisk-xip/files/asterisk.init new file mode 100755 index 0000000000..64a7d906ca --- /dev/null +++ b/contrib/asterisk-xip/files/asterisk.init @@ -0,0 +1,125 @@ +#!/bin/sh /etc/rc.common +# Copyright (C) 2006 OpenWrt.org +START=50 +STOP=50 + +DEST= +OPTIONS="" +DEFAULT=$DEST/etc/default/asterisk +UCILIB=$DEST/usr/lib/asterisk/uci +EXTRAPARAM=$1 + +export EXTRA_COMMANDS="console check down" +export EXTRA_HELP="\ + console Start asterisk console + check Test asterisk uci config + down Force asterisk to stop" + +reboot_ata() { + cd /tmp + wget -q http://ata.lan/admin/reboot -O - >&- 2>&- +} + +load_ucilib() . ${UCILIB}/asteriskuci + +start_uci() { + load_ucilib + + start_uci_asterisk $DEST +} +restart_uci() { + load_ucilib + + restart_uci_asterisk $DEST +} + +stop_uci() { + load_ucilib + + stop_uci_asterisk $DEST +} +reload_uci() { + load_ucilib + + reload_uci_asterisk "$DEST" +} + +start() { + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK-no} in + uci) start_uci ;; + yes) + [ -d /var/run ] || mkdir -p /var/run + [ -d $DEST/var/log/asterisk ] || mkdir -p $DEST/var/log/asterisk + [ -d $DEST/var/spool/asterisk ] || mkdir -p $DEST/var/spool/asterisk + [ -d /var/spool/asterisk ] || mkdir -p /var/spool/asterisk + [ -h $DEST/usr/lib/asterisk/astdb ] || ln -sf /var/spool/asterisk/astdb $DEST/usr/lib/asterisk/astdb + $DEST/usr/sbin/asterisk $OPTIONS -f 2>&1 > $DEST/var/log/asterisk/asterisk_proc & + ( sleep 5; reboot_ata ) & + ;; + *) return 1 ;; + esac +} + +stop() { + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK} in + uci) stop_uci ;; + *) [ -f /var/run/asterisk.pid ] && kill $(cat /var/run/asterisk.pid) 2>&- >&- + esac +} + +console() { + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK} in + uci) $DEST/usr/sbin/asterisk $UCIOPTIONS -C /tmp/asterisk/asterisk.conf -r ;; + yes) $DEST/usr/sbin/asterisk $OPTIONS -r ;; + esac + +} +check() { + load_ucilib + + setup_asterisk "$DEST" test "$EXTRAPARAM" +} + +reload() { + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK-no} in + uci) reload_uci ;; + yes) restart ;; + esac + +} + +restart() { + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK-no} in + uci) restart_uci ;; + yes) + if [ -r /var/run/asterisk.ctl ] ; then + if $DEST/usr/sbin/asterisk -r -x "restart gracefully" 2>&- >&- ; then + echo "Restarting when convenient" + return 0 + fi + fi + stop + start + esac +} + +down() { + if [ -r /var/run/asterisk.ctl ] ; then + [ -f $DEFAULT ] && . $DEFAULT + case ${ENABLE_ASTERISK} in + uci) $DEST/usr/sbin/asterisk -C /tmp/asterisk/asterisk.conf -r -x "stop now" 2>&- >&- ;; + *) $DEST/usr/sbin/asterisk $OPTIONS -r -x "stop now" 2>&- >&- + esac + [ -f /var/run/asterisk.pid ] && sleep 1 + fi + [ -f /var/run/asterisk.pid ] && kill $(cat /var/run/asterisk.pid) 2>&- >&- + [ -f /var/run/asterisk.pid ] && sleep 2 + [ -f /var/run/asterisk.pid ] && kill -9 $(cat /var/run/asterisk.pid) 2>&- >&- +} + +# vim:ts=2 sw=2 diff --git a/contrib/asterisk-xip/files/macros/clock.conf b/contrib/asterisk-xip/files/macros/clock.conf new file mode 100644 index 0000000000..3250b396a8 --- /dev/null +++ b/contrib/asterisk-xip/files/macros/clock.conf @@ -0,0 +1,23 @@ +; Talking clock by Michael Geddes +; Borrowed from bits here and there. +[macro-talkingclock] ; (TimeFormat, DateFormat, Zone) +exten => s,1,Answer +exten => s,n,set(tcTimeFormat=${ARG1}) +exten => s,n,GotoIf($["${tcTimeFormat}" = ""]?:tfOK) +exten => s,n,set(tcTimeFormat=HM\'vm-and\'S\'seconds\') +exten => s,n(tfOK),set(tcDateFormat=${ARG2}) +exten => s,n,GotoIf($["${tcDateFormat}" = ""]?:dfOK) +exten => s,n,set(tcDateFormat=AdBY) +exten => s,n(dfOK),set(tcZone=${ARG3}) +exten => s,n,GotoIf($["${tcZone}" = ""]?:znOK) +exten => s,n,set(tcZone=${TalkingClockZone}) +exten => s,n(znOK),SayUnixTime(,${tcZone},${tcDateFormat}) +exten => s,n(again),Wait(2) +exten => s,n,Set(FutureTime=$[${EPOCH} + 6]) +exten => s,n,Playback(misc/at-tone-time-exactly) +exten => s,n,SayUnixTime(${FutureTime},${tcZone},${tcTimeFormat}) +; Wait to say the beep. +exten => s,n(waitforit),noop +exten => s,n,GotoIf($[ ${EPOCH} < ${FutureTime} ]?waitforit:) +exten => s,n,playback(beep) +exten => s,n,goto(again) diff --git a/contrib/asterisk-xip/files/macros/lastcall.conf b/contrib/asterisk-xip/files/macros/lastcall.conf new file mode 100644 index 0000000000..5d178617fe --- /dev/null +++ b/contrib/asterisk-xip/files/macros/lastcall.conf @@ -0,0 +1,78 @@ +; Last-Called number storage and calling. +; Author: Michael Geddes aka FrogOnWheels + +[macro-lastcallstore] ; (Number , EntryType, BufferSize) +exten => s,1,set(lcsName=lastcall) +exten => s,n,set(lcsCount=10) +exten => s,n,GotoIf($["${ARG2}" = ""]?blankarg) +exten => s,n,GotoIf($["${ARG2}" = "lastcall"]?blankarg) +exten => s,n,Set(lcsName=lastcall_${ARG2}) +exten => s,n(blankarg),GotoIf($["${ARG3}" = ""]?nocount) +exten => s,n,Set(lcsCount=${ARG3}) +exten => s,n(nocount),Noop(${lcsName}:${DB(${lcsName}/number1)}:${ARG1}) +exten => s,n,GotoIf($["${DB(${lcsName}/number1)}" = "${ARG1}"]?setdate) +exten => s,n,set(CallerPointer=1) + +exten => s,n(again),GotoIf($["${DB(${lcsName}/ddate${CallerPointer})}" = ""]?copynext) +exten => s,n,GotoIf($["${DB(${lcsName}/number${CallerPointer})}" = "${ARG1}"]?copynext) +exten => s,n,Set(CallerPointer=$[${CallerPointer}+1]) +exten => s,n,GotoIf($[${CallerPointer} <= ${lcsCount}]?again) + +exten => s,n(copynext),set(DB(${lcsName}/ddate$[${CallerPointer}])=${DB(${lcsName}/ddate$[${CallerPointer}-1])}) +exten => s,n,set(DB(${lcsName}/number$[${CallerPointer}])=${DB(${lcsName}/number$[${CallerPointer}-1])}) +exten => s,n,set(CallerPointer=$[${CallerPointer}-1]) +exten => s,n,GotoIf($[${CallerPointer} > 0]?copynext) +exten => s,n,set(DB(${lcsName}/number1)=${ARG1}) +exten => s,n(setdate),set(DB(${lcsName}/ddate1)=${EPOCH}) + +[macro-lastcallapp] ; (Entrytype, Count, RingContext, Tag) +exten => s,1,set(lcsName=lastcall) +exten => s,n,set(lcsCount=10) +exten => s,n,GotoIf($["${ARG1}" = ""]?blankName) +exten => s,n,Set(lcsName=lastcall_${ARG1}) +exten => s,n(blankName),GotoIf($["${ARG2}" = ""]?nocount) +exten => s,n,Set(lcsCount=${ARG2}) +exten => s,n(nocount),set(lcsCallContext=internal) +exten => s,n,GotoIf($["${ARG3}" = ""]?blankContext) +exten => s,n,Set(lcsCallContext=${ARG3}) +exten => s,n(blankContext),set(lcsTag=${ARG4}) +exten => s,n,GotoIf($["${lcsTag}" != ""]?hasTag) +exten => s,n,Set(lcsTag=lastcall/previous-numbers) +exten => s,n(hasTag),Set(lcsPointer=1) +exten => s,n,GotoIf($["${DB(${lcsName}/ddate1)}" != ""]?macrobody_lastcallapp|s|1) +exten => s,n,playback(${lcsTag}&lastcall/none-available) +[macrobody_lastcallapp] +exten => s,1(repeat),background(${lcsTag}) +exten => s,n(again),wait(1) +exten => s,n,Set(lcsLastnum=${DB(${lcsName}/number${lcsPointer})}) +exten => s,n,Set(ddate=${DB(${lcsName}/ddate${lcsPointer})}) +exten => s,n,GotoIf($["${lcsLastnum}" != "anonymous"]?checkblank) +exten => s,n,Set(lcsLastnum="") +exten => s,n(checkblank),GotoIf($["${lcsLastnum}" = ""]?noinfo) +exten => s,n,saydigits(${lcsLastnum}) +exten => s,n(saycalltime),wait(.5) +exten => s,n,sayunixtime(${ddate},${LASTCALLZONE},QIMp) +exten => s,n(saymenu),background(silence/1) +exten => s,n,Set(lcsLastDate=${DB(${lcsName}/ddate$[ ${lcsPointer} + 1])}) +exten => s,n,GotoIf($[$[${lcsPointer} = ${lcsCount}] | $["${lcsLastDate}" = ""]]?noprev) +exten => s,n,background(lastcall/next) +exten => s,n(noprev),GotoIf($["${lcsLastnum}" = ""]?nocall) +exten => s,n,background(lastcall/call-number) +exten => s,n(nocall),GotoIf($[${lcsPointer} = 1]?nonext) +exten => s,n,background(lastcall/previous) +exten => s,n(nonext),background(silence/10) +exten => s,n,Goto(repeat) +exten => s,n(noinfo),background(lastcall/no-number-info) +exten => s,n,goto(saycalltime) +exten => 5,1,GotoIf($["${lcsLastnum}" = ""]?noinfo]) +exten => 5,n,Ringing() +exten => 5,n,Goto(${lcsCallContext},${lcsLastnum},1) +exten => 6,1,GotoIf($[$[${lcsPointer} = ${lcsCount}] | $["${lcsLastDate}" = ""]]?sayn) +exten => 6,n,Set(lcsPointer=$[${lcsPointer} + 1]) +exten => 4,1,GotoIf($[${lcsPointer}=1]?sayn) +exten => 4,n,Set(lcsPointer=$[${lcsPointer} - 1]) +exten => _[46],n(sayn),saynumber(${lcsPointer}) +exten => _[46],n,goto(s|again) +exten => i,1,Goto(s|again) +exten => t,1,playback(goodbye) +exten => t,n,Hangup diff --git a/contrib/asterisk-xip/files/modules.conf b/contrib/asterisk-xip/files/modules.conf new file mode 100644 index 0000000000..ce12c82dcd --- /dev/null +++ b/contrib/asterisk-xip/files/modules.conf @@ -0,0 +1,137 @@ +; +; Asterisk configuration file +; +; Module Loader configuration file +; + +[modules] +autoload=yes +; +; Any modules that need to be loaded before the Asterisk core has been +; initialized (just after the logger has been initialized) can be loaded +; using 'preload'. This will frequently be needed if you wish to map all +; module configuration files into Realtime storage, since the Realtime +; driver will need to be loaded before the modules using those configuration +; files are initialized. +; +; An example of loading ODBC support would be: +;preload => res_odbc.so +;preload => res_config_odbc.so +; +noload => res_config_mysql.so ; +noload => res_crypto.so ; Cryptographic Digital Signatures +; load => res_features.so ; Call Parking Resource +noload => res_indications.so ; Indications Configuration +noload => res_monitor.so ; Call Monitoring Resource +; load => res_musiconhold.so ; Music On Hold Resource +noload => cdr_csv.so ; Comma Separated Values CDR Backend +noload => cdr_custom.so ; Customizable Comma Separated Values CDR Backend +noload => cdr_manager.so ; Asterisk Call Manager CDR Backend +noload => cdr_mysql.so ; MySQL CDR Backend +noload => cdr_pgsql.so ; PostgreSQL CDR Backend +noload => cdr_sqlite.so ; SQLite CDR Backend +noload => chan_alsa.so ; Channel driver for GTalk +noload => chan_agent.so ; Agent Proxy Channel +noload => chan_gtalk.so ; Channel driver for GTalk +; load => chan_iax2.so ; Inter Asterisk eXchange (Ver 2) +; load => chan_local.so ; Local Proxy Channel +; load => chan_sip.so ; Session Initiation Protocol (SIP) +noload => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder +noload => codec_adpcm.so ; Adaptive Differential PCM Coder/Decoder +noload => codec_alaw.so ; A-law Coder/Decoder +noload => codec_g726.so ; ITU G.726-32kbps G726 Transcoder +; load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translation +; load => codec_ulaw.so ; Mu-law Coder/Decoder +noload => codec_speex.so ; Speex/PCM16 (signed linear) Codec Translator +noload => format_au.so ; Sun Microsystems AU format (signed linear) +noload => format_g723.so ; G.723.1 Simple Timestamp File Format +noload => format_g726.so ; Raw G.726 (16/24/32/40kbps) data +noload => format_g729.so ; Raw G729 data +; load => format_gsm.so ; Raw GSM data +noload => format_h263.so ; Raw h263 data +noload => format_jpeg.so ; JPEG (Joint Picture Experts Group) Image +; load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) +noload => format_pcm_alaw.so ; Raw aLaw 8khz PCM Audio support +noload => format_sln.so ; Raw Signed Linear Audio support (SLN) +noload => format_vox.so ; Dialogic VOX (ADPCM) File Format +; load => format_wav.so ; Microsoft WAV format (8000hz Signed Line +; load => format_wav_gsm.so ; Microsoft WAV format (Proprietary GSM) +noload => app_alarmreceiver.so ; Alarm Receiver Application +noload => app_authenticate.so ; Authentication Application +noload => app_cdr.so ; Make sure asterisk doesn't save CDR +noload => app_chanisavail.so ; Check if channel is available +noload => app_chanspy.so ; Listen in on any channel +noload => app_controlplayback.so ; Control Playback Application +noload => app_cut.so ; Cuts up variables +noload => app_db.so ; Database access functions +; load => app_dial.so ; Dialing Application +noload => app_dictate.so ; Virtual Dictation Machine Application +noload => app_directory.so ; Extension Directory +noload => app_directed_pickup.so ; Directed Call Pickup Support +noload => app_disa.so ; DISA (Direct Inward System Access) Application +noload => app_dumpchan.so ; Dump channel variables Application +; load => app_echo.so ; Simple Echo Application +noload => app_enumlookup.so ; ENUM Lookup +noload => app_eval.so ; Reevaluates strings +noload => app_exec.so ; Executes applications +noload => app_externalivr.so ; External IVR application interface +noload => app_forkcdr.so ; Fork The CDR into 2 seperate entities +noload => app_getcpeid.so ; Get ADSI CPE ID +noload => app_groupcount.so ; Group Management Routines +noload => app_ices.so ; Encode and Stream via icecast and ices +noload => app_image.so ; Image Transmission Application +noload => app_lookupblacklist.so ; Look up Caller*ID name/number from black +noload => app_lookupcidname.so ; Look up CallerID Name from local databas +; load => app_macro.so ; Extension Macros +noload => app_math.so ; A simple math Application +noload => app_md5.so ; MD5 checksum Application +; load => app_milliwatt.so ; Digital Milliwatt (mu-law) Test Application +noload => app_mixmonitor.so ; Record a call and mix the audio during the recording +noload => app_parkandannounce.so ; Call Parking and Announce Application +; load => app_playback.so ; Trivial Playback Application +noload => app_privacy.so ; Require phone number to be entered +noload => app_queue.so ; True Call Queueing +noload => app_random.so ; Random goto +noload => app_read.so ; Read Variable Application +noload => app_readfile.so ; Read in a file +noload => app_realtime.so ; Realtime Data Lookup/Rewrite +noload => app_record.so ; Trivial Record Application +; load => app_sayunixtime.so ; Say time +noload => app_senddtmf.so ; Send DTMF digits Application +noload => app_sendtext.so ; Send Text Applications +noload => app_setcallerid.so ; Set CallerID Application +noload => app_setcdruserfield.so ; CDR user field apps +noload => app_setcidname.so ; Set CallerID Name +noload => app_setcidnum.so ; Set CallerID Number +noload => app_setrdnis.so ; Set RDNIS Number +noload => app_settransfercapability.so ; Set ISDN Transfer Capability +noload => app_sms.so ; SMS/PSTN handler +noload => app_softhangup.so ; Hangs up the requested channel +noload => app_stack.so ; Stack Routines +noload => app_system.so ; Generic System() application +noload => app_talkdetect.so ; Playback with Talk Detection +noload => app_test.so ; Interface Test Application +noload => app_transfer.so ; Transfer +noload => app_txtcidname.so ; TXTCIDName +noload => app_url.so ; Send URL Applications +noload => app_userevent.so ; Custom User Event Application +; load => app_verbose.so ; Send verbose output +noload => app_waitforring.so ; Waits until first ring after time +noload => app_waitforsilence.so ; Wait For Silence Application +noload => app_while.so ; While Loops and Conditional Execution +noload => func_callerid.so ; Caller ID related dialplan functions +noload => func_enum.so ; ENUM Functions +noload => func_uri.so ; URI encoding / decoding functions +noload => pbx_ael.so ; Asterisk Extension Language Compiler +; load => pbx_config.so ; Text Extension Configuration +noload => pbx_functions.so ; Builtin dialplan functions +noload => pbx_loopback.so ; Loopback Switch +noload => pbx_realtime.so ; Realtime Switch +noload => pbx_spool.so ; Outgoing Spool Support +noload => pbx_wilcalu.so ; Wil Cal U (Auto Dialer) +; +; Module names listed in "global" section will have symbols globally +; exported to modules loaded after them. +; +[global] +chan_modem.so=no diff --git a/contrib/asterisk-xip/files/uci/asteriskconf b/contrib/asterisk-xip/files/uci/asteriskconf new file mode 100755 index 0000000000..d90f9d9cd9 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/asteriskconf @@ -0,0 +1,144 @@ +#!/bin/sh + +# Asterisk.conf + +init_asteriskconf() { + + ast_add_reload dialplan + ast_enable_type asterisk + ast_enable_type setglobal + ast_enable_type include + # ast_enable_type hardware + ast_enable_type hardwarereboot + + + asterisk_zone="Australia/Perth" + asterisk_spooldir="${DEST}/var/spool/asterisk" + asterisk_logdir="${DEST}/var/log/asterisk" + asterisk_agidir="${DEST}/usr/lib/asterisk/agi-bin" + return 0 +} + +asterisk_option_list="verbose debug quiet dontwarn timestamp execincludes \ +highpriority initcrypto nocolor dumpcore languageprefix internal_timing \ +systemname maxcalls maxload cache_record_files record_cache_dir \ +transmit_silence_during_record transcode_via_sln runuser rungroup" +asterisk_path_list="spooldir logdir agidir" + +valid_asterisk_option() { + is_in_list $1 ${asterisk_option_list} ${asterisk_path_list} zone + return $? +} + +create_asteriskconf() { + # echo ${DEST_DIR} + file=${DEST_DIR}/asterisk.conf + get_checksum asterisk_conf $file + + echo "${asteriskuci_gen}${N}[directories] +astetcdir => ${DEST_DIR} +astmoddir => ${DEST}/usr/lib/asterisk/modules +astvarlibdir => ${DEST}/usr/lib/asterisk +astdatadir => ${DEST}/usr/lib/asterisk +astrundir => /var/run" > $file + for i in ${asterisk_path_list} ; do + eval "value=\"\${asterisk_$i}\"" + if [ ! -z $value ] ; then + echo "ast$i => $value" >> $file + fi + done + echo "${N}[options]" >> $file + + for i in ${asterisk_option_list} ; do + eval "value=\"\${asterisk_$i}\"" + if [ ! -z $value ] ; then + echo "$i => $value" >> $file + fi + done + + echo "${N}; Changing the following lines may compromise your security. +[files] +astctlpermissions = 0660 +astctlowner = root +astctlgroup = nogroup +astctl = asterisk.ctl " >> $file + check_checksum "$asterisk_conf" "$file" || ast_restart=1 + +} + +handle_asterisk() { + option_cb() { + case $1 in + zone) + asterisk_zone="$2";; + *) + if valid_asterisk_option $1 $2 ; then + eval "asterisk_${1}=\"$2\"" + else + logerror "Invalid Asterisk option: $1" + fi + esac + } +} + +handle_include(){ + option_cb() { + case $1 in + dialplan|dialplan_ITEM*) + append dialplan_includes "#include <$2>" "${N}" + ;; + dialplan_COUNT) ;; + *) logerror "Invalid option \"$1\" for include" ;; + esac + } +} + +handle_setglobal() { + option_cb() { + case $1 in + set_COUNT) ;; + set|set_ITEM*) + if [ "${2%=*}" == "${2}" ] ; then + logerror "SetGlobal option \"$2\" not of the form VARIABLE=Value" + else + append dialplan_globals "" "${N}" + fi ;; + *) logerror "Invalid option \"$1\" for setglobal" ;; + esac + } +} + +handle_hardwarereboot() handle_hardware reboot + +# Handle hardware options (reboot) for Softphones +handle_hardware() { + case $1 in + reboot) + hardware_method= + hardware_param= + option_cb() { + case $1 in + method) + hardware_method="$2";; + param) + case ${hardware_method} in + web) append hardware_reboots "wget -q $2 -O - >&- 2>&-" "${N}" ;; + system) append hardware_reboots "$2 >&- 2>&-" "${N}" ;; + *) logerror "Invalid Hardware reboot method: ${hardware_method}" + esac + esac + + } + ;; + *) logerror "Invalid Hardware option: $1" + esac +} + +reboot_hardware() { + cd /tmp + eval ${hardware_reboots} +} + + + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/asteriskconf.txt b/contrib/asterisk-xip/files/uci/asteriskconf.txt new file mode 100644 index 0000000000..8966cb844a --- /dev/null +++ b/contrib/asterisk-xip/files/uci/asteriskconf.txt @@ -0,0 +1,41 @@ + +asterisk + zone - Default TimeZone + + Various asterisk.conf options + verbose + debug + quiet + dontwarn + timestamp + execincludes + highpriority + initcrypto + nocolor + dumpcore + languageprefix + internal_timing + systemname + maxcalls + maxload + cache_record_files + record_cache_dir + transmit_silence_during_record + transcode_via_sln + runuser + rungroup + + spooldir + logdir + agidir + +setglobal + set (list) - VARIABLE=Value Global Variables + +include + dialplan (list) - Files to #include + +hardwarereboot - Reboot phone hardware + method - web (wget), system + param - url/program for reboot + diff --git a/contrib/asterisk-xip/files/uci/asteriskuci b/contrib/asterisk-xip/files/uci/asteriskuci new file mode 100755 index 0000000000..1fd8f99b9d --- /dev/null +++ b/contrib/asterisk-xip/files/uci/asteriskuci @@ -0,0 +1,365 @@ +#!/bin/sh + +# Author: Michael Geddes <michael at frog dot wheelycreek dot net> +# Copyright 2008 Michael Geddes +# Licensed under GPL +Version=0.8 + + +# Todo +# Calling of Macros in dialplan +# Create a Menu +# Incoming Zones + +debuglevel=0 + +. /etc/functions.sh + +asteriskuci_gen="; Generated by Openwrt AstriskUCI script version ${Version}$N" + +# Utils + +logerror() { + echo "Error: $1" +} + +logdebug() { + if [ $(expr $1 "<=" ${debuglevel-0}) == 1 ] ; then + echo "Log: $2" + fi +} + +is_in_list(){ + val=$1 + shift + for i in $* ; do + [ $i == $val ] && return 0 + done + return 1 +} + +split_append() { + local lhs="$2" + local rhs="$3" + + while [ ! -z "$rhs" ] ; do + cur=${rhs%%,*} + nvar=${rhs#*,} + [ -z $4 ] || eval "$4 ${cur}" + append $1 "${lhs}${cur}" "$4" + [ "$nvar" == "$rhs" ] && break + rhs=${nvar} + done +} + +get_checksum() { + if [ -r "$2" ] ; then + local sum=`md5sum $2 | cut -d " " -f 1` + eval "$1=\"$sum\"" + else + eval "$1=NONE" + fi + #eval "logdebug 1 \"Checksum $2 : \${$1}\"" +} + +check_checksum() { + if [ -r "$2" ] ; then + local sum=`md5sum $2 | cut -d " " -f 1` + else + eval sum=NONE + fi + #logdebug 1 "Compare $1 checksum $2 with new checksum $sum " + [ "$sum" == "$1" ] + return $? +} + +# Add config module to initialise list +ast_add_conf() append asterisk_conf_list $1 " " +# Add module to initialise list +ast_add_module() append asterisk_module_list $1 " " +# Add to 'reload' list. +ast_add_reload() append asterisk_load_list $1 " " + +# Enable a top-level type +ast_enable_type() eval "enabled_section_${1}=1" + +# Is a top-level type enabled? +ast_type_enabled() { + eval "local res=\${enabled_section_${1}}" + if [ "$res" != 1 ] ; then + return 1 #Fail + fi + return 0 +} + +# For use in sections - make sure that the last section is processed +check_add() { + logdebug 1 "Check add $1" + if [ ! -z "${last_added_checked}" ] ; then + logdebug 1 "Eval check-add ${last_added_checked}" + eval "check_add_${last_added_checked}" + fi + last_added_checked=$1 +} + +# Process the section yet to be checked. +check_all_added() check_add "" + +# Create static links for stuff we dont want to configure yet. +create_staticlinks() { + logdebug 1 "Link in a few mostly static configurations" + linkconfigs="codecs.conf say.conf sip_notify.conf udptl.conf logger.conf" + module_enabled res_indications && append linkconfigs indications.conf " " + for i in ${linkconfigs} ; do + [ -e $DEST_DIR/$i ] || ln -s $DEST/etc/asterisk/$i $DEST_DIR + done + + logdebug 1 "Link in #include directories" + for i in include inc libs lib library macro macros ; do + if [ -e $DEST/etc/asterisk/$i -a ! -d "$DEST_DIR/$i" -a ! -e "$DEST_DIR/$i" ] ; then + ln -s $DEST/etc/asterisk/$i $DEST_DIR + fi + done +} + +# default reboot +reboot_hardware() {} + + +# Top level handler +setup_asterisk() { + DEST=${1%/} + DEST_DIR=/tmp/asterisk + + testing_mode=0 + if [ "$2" == "testonly" ] ; then + testing_mode=1 + elif [ "$2" == "test" ] ; then + DEST_DIR=/tmp/asterisk.tmp + echo Using Test dir: $DEST_DIR + testing_mode=2 + fi + + [ -z "$3" ] || debuglevel=$3 + + logdebug 1 "Loading Asterisk Config" + . ${UCILIB}/asteriskconf + logdebug 2 "Loading Module Config" + . ${UCILIB}/moduleconf + logdebug 2 "Loading Dialplan Config" + . ${UCILIB}/dialplanconf + + for f in ${DEST}/etc/asterisk/conf.d/* ; do + logdebug 1 "Loading Module $f" + [ -f $f ] && . $f + done + + include /lib/network + scan_interfaces + + init_asteriskconf + init_moduleconf + init_dialplanconf + + for i in ${asterisk_module_list} ; do + logdebug 1 "Init $i module" + eval "init_${i}" + done + + for i in ${asterisk_conf_list} ; do + logdebug 1 "Init $i config" + eval "init_${i}conf" + done + + config_cb() { + cur_section=$1/$2 + logdebug 2 "Load $1/$2" + eval "local val=\"\${dups_$2}\"" + if [ "${val}" == "" ] ; then + eval "dups_$2=1" + else + logerror "Duplicate Section Name: $2 (type $1)" + fi + + if ast_type_enabled $1 ; then + eval "handle_$1 \$2" + elif [ ! -z "$1" ] ; then + + logerror "Unknown section: $1/$2" + option_cb() { + logerror "Invalid option '$1' for invalid section" + } + fi + } + config_load asterisk + check_all_added + + if [ "$testing_mode" != "1" ] ; then + mkdir -p ${DEST_DIR} + + create_asteriskconf + for i in ${asterisk_conf_list} ; do + logdebug 1 "Create $i config" + eval "create_${i}conf" + done + create_dialplanconf + create_moduleconf + + # Link in a few mostly static configurations + create_staticlinks + fi + [ "$testing_mode" == "2" ] && reload_check_asterisk + return 0 +} + +astcmd() { + ASTCMD="${DEST%/}/usr/sbin/asterisk -C /tmp/asterisk/asterisk.conf " + logdebug 1 "Command: $1" + if [ -z "${2-}" ] ; then + ${ASTCMD} -r -x "$1" 2>&- 1>&- + else + eval "$2=`${ASTCMD} -r -x \"$1\"`" + fi + return $? +} + +# waitfor() { +# while [ -d /proc/$1 ] ; do +# sleep 1 +# done +# } + +restart_gracefully() { + stop_uci_asterisk "$DEST" + startup_asterisk "$DEST" + #ret=0 + #echo "Check for pid" + #if [ -r /var/run/asterisk.ctl ] ; then + # astcmd "stop gracefully" + # local ret=$? + # [ ${ret} = 0 ] || return $ret + # waitfor `cat /var/run/asterisk.pid` + #fi + #startup_asterisk ${DEST} + return 0 +} +astcmds() { + while [ ! -z "$1" ] ; do + astcmd "$1" + shift + done +} + +reload_check_asterisk() { + logdebug 1 "Check Reloading" + local reboot=0 + if [ "${ast_restart-}" == 1 ] ; then + logdebug 1 "Restarting Gracefully" + reboot=0 + else + for i in ${asterisk_load_list} ; do + logdebug 1 "Checking ${i} reload" + eval "local doload=\${ast_${i}_restart}" + case $doload in + 1) logdebug 1 "Reloading ${i}" ;; + 2) logdebug 1 "Unloading ${i}" ;; + esac + done + fi + [ ${reboot} = 1 ] && logdebug 1 "reboot hardware" +} + +reload_asterisk() { + logdebug 1 "Reloading" + local reboot=0 + if [ "${ast_restart-}" == 1 ] ; then + logdebug 2 "Restarting Gracefully" + restart_gracefully + reboot=0 + else + for i in ${asterisk_load_list} ; do + logdebug 3 "Checking ${i} reload" + eval "local doload=\${ast_${i}_restart}" + case $doload in + 1) logdebug 1 "Reloading ${i}" + eval "reload_${i}" || reboot=1 ;; + 2) logdebug 1 "Unloading ${i}" + eval "unload_${i}" || reboot=1 ;; + esac + done + fi + + if [ ${reboot} = 1 ] ; then + ( sleep 5; reboot_hardware ) & + fi +} +startup_asterisk() { + DEST="${1%/}" + DEFAULT=$DEST/etc/default/asterisk + [ -f $DEFAULT ] && . $DEFAULT + [ -d /var/run ] || mkdir -p /var/run + [ -d ${asterisk_logdir} ] || mkdir -p ${asterisk_logdir} + [ -d ${asterisk_spooldir} ] || mkdir -p ${asterisk_spooldir} + [ -d /var/spool/asterisk ] || mkdir -p /var/spool/asterisk + [ -h $DEST/usr/lib/asterisk/astdb ] || ln -sf /var/spool/asterisk/astdb $DEST/usr/lib/asterisk/astdb + + $DEST/usr/sbin/asterisk -C /tmp/asterisk/asterisk.conf $UCIOPTIONS -f 2>&1 > ${asterisk_logdir}/asterisk_proc & + # Wait a bit then reboot the hardware + ( sleep 5; reboot_hardware ) & +} + +# Init.d start() handler +start_uci_asterisk() { + DEST="${1%/}" + + if setup_asterisk $DEST ; then + startup_asterisk "$DEST" + fi +} + +restart_uci_asterisk() { + DEST="${1%/}" + if setup_asterisk $DEST ; then + echo "Trying to Restart gracefully" + if [ -r /var/run/asterisk.ctl ] ; then +# if astcmd "restart gracefully" ; then + echo "Sending restart" + if restart_gracefully ; then + echo "Restarting gracefully" + return 0 + fi + fi + stop_uci_asterisk "$DEST" + startup_asterisk "$DEST" + else + stop_uci_asterisk $1 + echo "Setup Failed" + return 1 + fi +} + +# init.d stop() handler +stop_uci_asterisk() { + DEST=${1%/} + if [ -r /var/run/asterisk.ctl ] ; then + astcmd "stop now" + sleep 1 + fi + [ -f /var/run/asterisk.pid ] && kill $(cat /var/run/asterisk.pid) >/dev/null 2>&1 +} + +reload_uci_asterisk() { + DEST=${1%/} + DEFAULT=$DEST/etc/default/asterisk + + if [ -r /var/run/asterisk.ctl ] ; then + if setup_asterisk "$DEST" ; then + # Selective reload modules. + reload_asterisk + fi + else + start_uci_asterisk "$1" + fi +} + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/dialplanconf b/contrib/asterisk-xip/files/uci/dialplanconf new file mode 100755 index 0000000000..70ef7c5460 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/dialplanconf @@ -0,0 +1,742 @@ +#!/bin/sh + +# Dialplans (extensions.conf) + +# Implicit: ast_add_conf dialplan +init_dialplanconf() { + ast_enable_type dialplangeneral + ast_enable_type dialplan + ast_enable_type dialplanexten + ast_enable_type dialplangoto + ast_enable_type dialplansaytime + ast_enable_type dialzone + ast_enable_type inzone + ast_enable_type incominggeneral + ast_enable_type incoming + + dialplan_allowtransfer=no + dialplan_dialtimeout=30 + dialplan_answerfirst=no + dialplan_static=yes + dialplan_writeprotect=no + dialplan_canreinvite=no + dialplan_includes= + dialplan_globals= + return 0 +} + +dialplangeneral_list="static writeprotect canreinvite clearglobalvars" +dialplangeneral_list_ex="lastdialed lastdialedtype voiceboxext answerfirst dialtimeout allowtransfer international internationalout" + +valid_dialplangeneral() { + for i in ${dialplangeneral_list} ${dialplangeneral_list_ex} ; do + [ "$i" == "$1" ] && return 0 + done + return 1 +} + + +check_add_context() { + local context="${1}" + local check="${context#macro-}" + [ "${context}" == ${check} ] || check="macro__${check}" + eval "local isadded=\"\${dialplan_add_context_${check}-0}\"" + if [ "$isadded" != "1" ] ; then + eval "dialplan_add_context_${check}=1" + append dialplan_contexts "$context" + return 0 + else + return 1 + fi +} +append_dialplan_context() { + local context="${1}" + local check="${context#macro-}" + [ "${context}" == ${check} ] || check="macro__${check}" + append dialplan_context_${check} "${2}" "${N}" +} + +reload_dialplan() astcmd "dialplan reload" + +# Voicemail + +enable_voicemail() { + enable_module res_adsi + enable_module app_voicemail + enable_format gsm +} + +add_dialplan_exten() { + local context=$1 + logdebug 3 "Exten: $2" + local ext="exten => $2," + local planopt= + local timeout=${dialplan_dialtimeout} + # Answer extensions first. + local answerfirst=${dialplan_answerfirst} + local mailbox=$4 + [ -z "$5" ] || timeout=$5 + [ -z "$6" ] || answerfirst=$6 + + check_add_context "$context" + + if [ "$dialplan_allowtransfer" == "yes" ] ; then + planopt=${planopt}t + fi + if [ ! -z "${planopt}" ] ; then + planopt=",${timeout},${planopt}" + elif [ ! -z "${timeout}" ] ; then + planopt=",${timeout}" + fi + local dial="Dial($3$planopt)" + local item="1," + if [ "$answerfirst" == "yes" ] ; then + append_dialplan_context ${context} "${ext}1,Answer" + item="n," + fi + append_dialplan_context ${context} "${ext}${item}${dial}" + if [ ! -z "${mailbox}" ] ; then + enable_voicemail + append_dialplan_context ${context} "${ext}n,VoiceMail(${mailbox})" + fi + + append_dialplan_context ${context} "${ext}n,Congestion" +} + +add_dialplan_include() { + local context=$1 + logdebug 1 "Adding Dialplan Include $1 $2" + check_add_context "$context" + + split_append dialplan_context_${context} "include => " "$2" "${N}" +} + +add_dialplan_saytime() { + local context=$1 + logdebug 1 "Adding Dialplan saytime $1 $2" + check_add_context "$context" + local ext="exten => $2," + if [ "$dialplan_add_context_saytime" != 1 ] ; then + append dialplan_contexts saytime " " + dialplan_add_context_saytime=1 + enable_format gsm + enable_module app_sayunixtime + local zone=${asterisk_zone} + [ ! -z "$3" ] && zone="$3" + local format="IMp AdbY" + [ ! -z "$4" ] && format="$4" + append dialplan_context_saytime "exten => s,1,SayUnixTime(,${zone},${format})" "${N}" + fi + append_dialplan_context ${context} "${ext}1,Goto(saytime,s,1)" +} + +add_dialplan_goto() { + local context=$1 + logdebug 1 "Adding Dialplan goto $1 $2 $3" + check_add_context "$context" + append dialplan_context_${context} "exten => $2,1,Goto($3,\${EXTEN},1)" "${N}" +} + +handle_inzone() { + # TODO - Incoming zones. + return 0 +} + +#generate_inzone() { + #[local_Vista] + #exten => _X.,1,Ringing() + #exten => _X.,n,Dial(SIP/Nokia,5) + #exten => _X.,n,Answer + #exten => _X.,n,wait(1) + #exten => _X.,n,Set(EXITCONTEXT=xyzzy) + #exten => _X.,n,Dial(SIP/Nokia,15,rd) + #exten => _X.,n,Hangup + #[xyzzy] + #exten => 1,1,Noop(xyzzy) + #exten => 1,n,SayAlpha(b) + #exten => 1,n,Hangup +#} + +append_dialplan_dialzone() { + local file=$1 + + # Add the dialzone contexts + logdebug 1 "Dialplan: Add the dialzone contexts" + for zonename in ${dzones_match} ; do + eval "local diallist=\${dzone_${zonename}_match-}" + echo "${N}[${zonename}]" >> $file + eval "dialz=\${dzone_match_use_${zonename}-}" + + for v in prefix internationalprefix alwaysinternational countrycode ; do + eval "local $v=\${target_${v}_${dialz}:-}" + eval logdebug 3 "\"${v} = '\${$v}'\"" + done + + for v in localzone addprefix localprefix; do + eval "local $v=\${dzone_${zonename}_${v}:-}" + done + while [ ! -z "$diallist" ] ; do + cur=${diallist%%,*} + nvar=${diallist#*,} + if [ "${alwaysinternational}" = "yes" ] ; then + # Always dial international number with this target + # remove 'localprefix' (usually 0) from 'addprefix' + logdebug 3 "Removing ${localprefix} from ${addprefix}" + addprefix=${addprefix#$localprefix} + local curlen=`expr length "${localprefix}"` + if [ $curlen != 0 ] ; then + # remove 0 (or local prefix) + echo "exten => _${localprefix}${cur},1,Goto(${dialz}_dial,${countrycode}${addprefix}\${EXTEN:$curlen},1)" >> $file + fi + echo "exten => _${cur},1,Goto(${dialz}_dial,${countrycode}${addprefix}\${EXTEN},1)" >> $file + else + echo "exten => _${cur},1,Goto(${dialz}_dial,${addprefix}\${EXTEN},1)" >> $file + fi + [ "$nvar" == "$diallist" ] && break + diallist=${nvar} + done + eval "local diallist=\${dzone_${zonename}_international-}" + if [ ! -z "${diallist}" ] ; then + logdebug 2 "International: ${diallist}" + while [ ! -z "$diallist" ] ; do + cur=${diallist%%,*} + nvar=${diallist#*,} + + local curlen=`expr length ${cur}` + if [ "$alwaysinternational" = "yes" ] ; then + echo "exten => _${cur},1,Goto(${dialz}_dial,${addprefix}\${EXTEN:${curlen}},1)" >> $file + else + echo "exten => _${cur}.,1,Goto(${zonename}_check,${addprefix}\${EXTEN:${curlen}},1)" >> $file + fi + [ "$nvar" == "$diallist" ] && break + diallist=${nvar} + done + + if [ "$alwaysinternational" != "yes" ] ; then + # Check for local country code + echo "[${zonename}_check]" >> $file + + local locallen=`expr length ${countrycode}` + echo "exten => _${countrycode}X.,1,Goto(${localzone-default},${localprefix-0}\${EXTEN:${locallen}},1)" >> $file + echo "exten => _X.,1,Goto(${dialz}_dial,${internationalprefix}\${EXTEN},1)" >> $file + fi + fi + done + logdebug 1 "Dialplan: Finish the dialzone contexts" + +} + +append_dialplan_dialzone_out(){ + local file=$1 + + # Add the dialzone target contexts (dialing out) + logdebug 1 "Dialplan: Add the dialzone target contexts" + for contype in SIP IAX ; do + eval local conlist=\${dzones_${contype}-} + logdebug 1 "Adding ${contype} targets: ${conlist}" + for conname in $conlist ; do + echo "${N}[${contype}_${conname}_dial]" >> $file + for v in prefix internationalprefix alwaysinternational countrycode timeout lastdialed lastdialedtype ; do + eval "local $v=\${target_${v}_${contype}_${conname}:-}" + done + + if [ -z "${lastdialed}" ] ; then + lastdialed=${dialplan_lastdialed} + lastdialedtype=${dialplan_lastdialedtype} + fi + + # [ -z "${lastcallout}" ] && lastcallout=${feature_lastcall_outqueue} + # if [ ! -z "${lastcalloutexten}" ] ; then + # eval "local added=\${lastcallout_outqueue_extension_${lastcalloutexten}}" + # if [ ! "${added}" == 1 ] ; then + # add_dialplan_lastcall extensions "${lastcalloutexten}" "${lastcallout}" "${feature_lastcall_outcount}" + # eval "local lastcallout_outqueue_extension_${lastcalloutexten}=1" + # fi + # fi + local ext="exten => _X.," + local en="1," + # Do not use prefix unles 'alwaysinternational' is yes + [ "$alwaysinternational" != "yes" ] && internationalprefix= + + # if [ ! -z "${lastcallout}" ] ; then # Add lastcall out + # enable_lastcall + # echo "${ext}${en}Macro(lastcallstore,${internationalprefix}\${EXTEN},${lastcallout},${feature_lastcall_outcount})" >> $file + # en="n," + # fi + + if [ ! -z "${lastdialed}" ] ; then + enable_module func_callerid + local lastparam="\${EXTEN}" + local lastmacro=${lastdialed} + local lastmacrotype=${lastdialedtype} + [ -z ${lastdialedtype} ] && lastdialedtype=goto + + local jumpmacroline= + gen_jumpmacro jumpmacroline "${ext}${en}" "" "${lastmacrotype}" "${lastmacro}" "${lastparam}" + echo ${jumpmacroline} >> $file + en="n," + fi + echo "${ext}${en}Dial(${contype}/${prefix}${internationalprefix}\${EXTEN}@${conname},$timeout)" >> $file + echo "exten => _X.,n,Congestion" >> $file + done + done +} + + +gen_jumpmacro() { + logdebug 3 "Gen JumpMacro /$1/=/$2/$3/$4/$5/$6/" + local leader=$2 + local condition=$3 + local macrotype=$4 + local name=$5 + local param=$6 + + if [ -z "${condition}" ] ; then + local cond="(" + else + local cond="If(${condition}?" + fi + case ${macrotype} in + gosub) + enable_module app_stack # for gosub + logdebug 1 "${1}=\"\${leader}Gosub\${cond}\${name},\${param},1)\"" + eval "${1}=\"\${leader}Gosub\${cond}\${name},\${param},1)\"" ;; + gosub_s) + enable_module app_stack # for gosub + logdebug 1 "${1}=\"\${leader}Gosub\${cond}\${name},s,1(\${param}))\"" + eval "${1}=\"\${leader}Gosub\${cond}\${name},s,1(\${param}))\"" ;; + macro) + enable_module app_macro + logdebug 1 "${1}=\"\${leader}Macro\${cond}\${name},\${param})\"" + eval "${1}=\"\${leader}Macro\${cond}\${name},\${param})\"" ;; + + s|start) + enable_module app_goto + logdebug 1 "${1}=\"\${leader}Goto(\${iz_target},s,1)\"" + eval "${1}=\"\${leader}Goto(\${iz_target},s,1)\"" ;; + *) + enable_module app_goto + logdebug 1 "${1}=\"\${leader}Goto\${cond}\${name},\${param},1)\"" + eval "${1}=\"\${leader}Goto\${cond}\${name},\${param},1)\"" ;; + esac +} + +append_jumpmacro(){ + local context=$1 + local jumpmacroline="" + gen_jumpmacro "jumpmacroline" "$2" "$3" "$4" "$5" "$6" + append_dialplan_context ${context} "${jumpmacroline}" +} + +append_dialplan_incoming(){ + local file=$1 + # Evaluate the incoming ringing dialplans + logdebug 1 "Add the 'incoming' dialplans: ${dialplan_extensions_incoming}" + for context in ${dialplan_extensions_incoming} ; do + eval "local curext=\"\${dialplan_incoming_$context}\"" + logdebug 2 "Adding incoming ${curext}" + + check_add_context "$context" + # lastcall lastcallexten lastmissed lastmissedexten + for i in answerfirst beforeanswer timeout menucontext \ + lastcall lastcalltype missed missedtype allowtransfer mailbox match matchcaller aftertimeout aftertimeouttype; do + eval "local iz_$i=\"\${incoming_${context}_$i}\"" + eval "logdebug 1 \"Incoming \$context: iz_\$i=\${iz_$i}\"" + done + [ ! -z ${iz_menucontext} ] && iz_answerfirst=yes + + if [ ! -z ${curext} ] ; then + [ -z ${iz_answerfirst} ] && iz_answerfirst=${incoming_answerfirst} +# [ -z ${iz_lastcall} ] && iz_lastcall=${feature_lastcall_inqueue} + if [ -z ${iz_lastcall} ] ; then + iz_lastcall=${incoming_lastcall} + iz_lastcalltype=${incoming_lastcalltype} + fi + if [ -z ${iz_missed} ] ; then + iz_missed=${incoming_missed} + iz_missedtype=${incoming_missedtype} + fi + [ -z ${iz_mailbox} ] && iz_mailbox=${incoming_mailbox} + [ -z ${iz_timeout} ] && iz_timeout=${incoming_timeout} + [ -z ${iz_allowtransfer} ] && iz_allowtransfer=${incoming_allowtransfer} + fi + + [ -z ${iz_match} ] && iz_match=_X. + [ ! -z ${iz_matchcaller} ] && iz_match=${iz_match}/${iz_matchcaller} + + local ext="exten => ${iz_match}," + local planopt= + [ "${iz_allowtransfer}" == "yes" ] && planopt=${planopt}t + local item="1," + + #append_dialplan_context ${context} "${ext}${item}Ringing()" + if [ ! -z "${iz_lastcall}" ] ; then + + enable_module func_callerid + local lastparam="\${CALLERID(num)}" + local lastmacrotype="${iz_lastcalltype}" + [ -z "${iz_lastcalltype}" ] && lastmacrotype=goto + local lastmacro=${iz_lastcall} + append_jumpmacro "${context}" "${ext}${item}" "" "${lastmacrotype}" "${lastmacro}" "${lastparam}" + item="n," + fi + if [ ! -z "${iz_missed}" ] ; then + enable_module func_callerid + local missedparam="\${CALLERID(num)}" + [ -z "${iz_missedtype}" ] && iz_missedtype=goto + + append_dialplan_context ${context} "${ext}${item}Set(lcsMissed=\${CALLERID(num)})" + item="n," + fi + # Ring before answering + if [ ! -z "${iz_beforeanswer}" -a ! -z "${curext}" ] ; then + append_dialplan_context ${context} "${ext}${item}Dial(${curext},${iz_beforeanswer},${planopt})" + iz_answerfirst=yes + item="n," + fi + # Now answer + if [ "$iz_answerfirst" == "yes" ] ; then + append_dialplan_context ${context} "${ext}${item}Answer" + item="n," + fi + # if [ ! -z ${iz_lastcallexten} ] ; then + # enable_lastcall + # add_dialplan_lastcall extensions "${iz_lastcallexten}" "${iz_lastcall}" "${feature_lastcall_incount}" + # fi + + if [ ! -z ${curext} ] ; then + if [ ! -z ${iz_menucontext} ] ; then + planopt=${planopt}rd + enable_module res_indications + # wait so the next ring connects. + append_dialplan_context ${context} "${ext}${item}Wait(1)" + append_dialplan_context ${context} "${ext}n,Set(EXITCONTEXT=${iz_menucontext})" + fi + + if [ ! -z ${planopt} ] ; then + planopt=",${iz_timeout-},${planopt}" + elif [ ! -z ${iz_timeout-} ] ; then + planopt=",${iz_timeout-}" + fi + local dial="Dial(${curext}${planopt})" + + append_dialplan_context ${context} "${ext}n,${dial})" + + if [ ! -z ${iz_missed} ] ; then + # It went to voicemail or was hung up - consider missed + append_jumpmacro ${context} "${ext}${item}" "" "${iz_missedtype}" "${iz_missed}" "\${lcsMissed}" + fi + fi + + local need_hangup=1 + # Add a goto or macro at the end + if [ ! -z ${iz_target} ] ; then + case ${iz_aftertimeouttype} in + macro) append_dialplan_context ${context} "${ext}${item}Macro(${iz_target})";; + s|start) append_dialplan_context ${context} "${ext}${item}Goto(${iz_target},s,1)" + need_hangup=0;; + *) append_dialplan_context ${context} "${ext}${item}Goto(${iz_target},\${EXTEN},1)" + need_hangup=0;; + esac + fi + + if [ ! -z ${iz_mailbox} ] ; then + enable_voicemail + append_dialplan_context ${context} "${ext}n,VoiceMail(${iz_mailbox})" + fi + + [ "${need_hangup}" = "1" ] && append_dialplan_context ${context} "${ext}n,Hangup" + + if [ ! -z ${iz_missed} ] ; then + # Check for missed call + + append_jumpmacro ${context} "exten => h,1," "\$[\"\${DIALSTATUS}\" = \"CANCEL\"]" "${iz_missedtype}" "${iz_missed}" "\${lcsMissed}" + #append dialplan_context_${context} \ + # "exten => h,1,MacroIf(\$[\"\${DIALSTATUS}\" = \"CANCEL\"]?lastcallstore,\${lcsMissed},${iz_lastmissed},${feature_lastcall_incount})" "${N}" + # [ ! -z ${iz_lastmissedexten} ] && \ + # add_dialplan_lastcall extensions "${iz_lastmissedexten}" "${iz_lastmissed}" "${feature_lastcall_incount}" + fi + done +} + +append_dialplan_extensions(){ + local file=$1 + # Evaluate the 'extension' dialplans + logdebug 1 "Add the 'extension' dialplans: ${dialplan_exts}" + for i in ${dialplan_exts} ; do + eval "local curext=\"\${dialplan_ext_$i}\"" + add_dialplan_exten extensions $i $curext + done +} + +append_include() { + append dialplan_includes "#include <$1>" "$N" +} + + +create_dialplanconf() { + # Add general section + logdebug 1 "Dialplan: Add general section" + local file=${DEST_DIR}/extensions.conf + get_checksum dialplan_conf $file + + echo "${asteriskuci_gen}" > $file + + [ -z "${dialplan_includes}" ] || (echo "${dialplan_includes}${N}" >> $file) + + if [ ! -z "${dialplan_globals}" ] ; then + echo "[globals]${N}${dialplan_globals}${N}" >> $file + fi + + echo "[general]" >> $file + for i in ${dialplangeneral_list} ; do + eval "local val=\"\$dialplan_$i\"" + if [ ! -z "$val" ] ; then + echo "$i=$val" >> $file + fi + done + + append_dialplan_dialzone "$file" + + append_dialplan_dialzone_out "$file" + + append_dialplan_park "$file" + + append_dialplan_extensions "$file" + + append_dialplan_incoming "$file" + + # append_dialplan_lastcall "$file" + + # Add all the contexts + logdebug 1 "Dialplan: Add all the contexts" + for i in ${dialplan_contexts} ; do + echo "${N}[$i]" >> $file + [ "${i#macro-}" == "${i}" ] || i=macro__${i#macro-} + eval "local curcontext=\"\${dialplan_context_$i-}\"" + echo "$curcontext">> $file + eval unset dialplan_context_$i + done + + check_checksum "$dialplan_conf" "$file" || ast_dialplan_restart=1 + + return 0 +} + +handle_dialplangeneral() { + option_cb(){ + if valid_dialplangeneral $1 $2 ; then + eval "dialplan_$1=\"$2\"" + else + logerror "Invalid Dialplan General option: $1" + fi + } +} + + +handle_dialplan(){ + context_name=$1 + if [ -z ${context_name} ] ; then + logerror "Context name required for dialplan" + context_name=default + fi + option_cb() { + logdebug 4 "dialplan ${context_name}: '$1' '$2'" + case $1 in + include_LENGTH) ;; + include|include_ITEM*) add_dialplan_include ${context_name} $2 ;; + *) + lhs=$1 + logdebug 4 "Add extension $lhs" + if [ "$(expr $lhs : [0-9][0-9]\*)" != 0 ] ; then + addtype=${2%%:*} + if [ "${addtype}" == "$2" ]; then + addparam= + else + addparam=${2#*:} + fi + case ${addtype} in + mailbox|voice) + add_dialplan_voice ${context_name} $1 $addparam ;; + meetme|conf|conference) add_dialplan_meetme ${context_name} $1 $addparam ;; + saytime|time) add_dialplan_saytime ${context_name} $1 $addparam ;; + clock) add_dialplan_talkclock ${context_name} $1 $addparam ;; + *) logerror "Unknown type '${addtype}' in dialplan ${context_name}, extension ${1}" + esac + else + logerror "Invalid option: $1 for dialplan ${context_name}" + fi + esac + } +} + +handle_dialplanexten() { + check_add dialplanexten + option_cb() { + case $1 in + dialplan|extension|type|target|dialextension|voicebox|timeout|answerfirst) + eval "dial_exten_$1=\"$2\"" + esac + } +} + +check_add_dialplanexten() { + if [ ! -z "${dial_exten_dialplan}" -a ! -z "${dial_exten_extension}" ] ; then + + local dialtarget=${dial_exten_type}/${dial_exten_dialextension+@}${dial_exten_dialextension}${dial_exten_target} + + check_add_context ${dial_exten_dialplan} + add_dialplan_exten "${dial_exten_dialplan}" "${dial_exten_extension}" \ + "${dialtarget}" "${dial_exten_voicebox}" "${dial_exten_timeout}" \ + "${dial_exten_answerfirst}" + fi + for i in dialplan extension voicebox type target dialextension timeout answerfirst ; do + eval "unset dial_voice_$i" + done +} + +handle_dialplansaytime() { + check_add dialplansaytime + option_cb() { + case $1 in + dialplan|extension|zone|format) + eval "dial_saytime_$1=\"$2\"" + esac + } +} + +check_add_dialplansaytime() { + logdebug 1 "Add Saytime to $1 / $2" + if [ ! -z "${dial_saytime_dialplan}" -a ! -z "${dial_saytime_extension}" ] ; then + # local ext="exten => ${dial_saytime_extension}," + [ -z "${dial_saytime_dialplan}" ] && dial_saytime_dialplan=default + + add_dialplan_saytime "${dial_saytime_dialplan}" "${dial_saytime_extension}" \ + "${dial_saytime_zone}" "${dial_saytime_format}" + fi + for i in dialplan extension zone format; do + eval "unset dial_saytime_$i" + done + +} + +handle_dialzone_match() { + eval "local isadded=\"\${dzone_${zone_name}-0}\"" + if [ "${isadded}" != "1" ] ; then + eval "dzone_${zone_name}=1" + append dzones_match ${zone_name} " " + fi + append dzone_${zone_name}_match $1 "," +} + +# Set up outgoing zones (match patterns) +handle_dialzone() { + zone_name=$1 + logdebug 2 "Loading dialzone: $zone_name" + option_cb(){ + logdebug 2 "Dialzone $1/$2" + case $1 in + uses) + local areatype=${2%[-/]*} + local areaname=${2#*[-/]} + logdebug 3 "Added: $areatype $areaname" + eval "local isadded=\"\${dzone_${areatype}_${areaname}-0}\"" + if [ "${isadded}" != "1" ] ; then + eval dzone_${areatype}_${areaname}=1 + append dzones_${areatype} "${areaname}" " " + fi + eval "dzone_match_use_${zone_name}=${areatype}_${areaname}" + logdebug 3 "Finished uses" + ;; + match_LENGTH) ;; + match|match_ITEM*) + handle_dialzone_match "$2" + ;; + international_LENGTH) ;; + international|international_ITEM*) + eval "local isadded=\"$dzone_${zone_name}\"" + if [ "${isadded}" != "1" ] ; then + eval "dzone_${zone_name}=1" + append dzones_match ${zone_name} " " + fi + append dzone_${zone_name}_international $2 "," + ;; + countrycode|localzone|addprefix|localprefix) + eval "dzone_${zone_name}_${1}=\"$2\"" + eval "y=\${dzone_${zone_name}_${1}}" + ;; + *) + logerror "Invalid Dialzone option: $1 in zone '${zone_name}'" ;; + esac + } +} + +check_add_dialplangoto() { + logdebug 1 "Add Goto to $1 / $2" + if [ ! -z "${dial_goto_dialplan}" -a ! -z "${dial_goto_extension}" ] ; then + local ext="exten => ${dial_goto_extension}," + check_add_context ${dial_goto_dialplan} + append_dialplan_context ${dial_goto_dialplan} "${ext}1,Goto(${dial_goto_target})" + fi + for i in dialplan extension room ; do + eval "unset dial_goto_$i" + done +} + +handle_dialplangoto(){ + check_add dialplangoto + option_cb() { + case $1 in + dialplan|extension|target) + eval "dial_goto_$1=\"$2\"" ;; + *) logerror "Invalid dialplangoto option: $1" + esac + } +} + +# Options for incoming calls. + +valid_incoming_list="allowtransfer timeout answerfirst mailbox lastcall lastcalltype missed missedtype" + +valid_incoming_option() { + is_in_list $1 ${valid_incoming_list} + return $? +} + +handle_incominggeneral() { + option_cb() { + if valid_incoming_option $1 ; then + logdebug 3 "Add Incoming General option $1=$2" + eval "incoming_${1}=\"$2\"" + else + logerror "Invalid incominggeneral option $1" + fi + } +} + +handle_incoming() { + incoming_context=$1 + incoming_list= + add_incoming_context "${incoming_context}" + + option_cb() { + logdebug 1 "Incoming ${incoming_context} $1=$2" + case $1 in + lastcall|lastcalltype|missed|missedtype) eval "incoming_${incoming_context}_${1}=\"$2\"" ;; +# lastcall|lastcallexten|lastmissed|lastmissedexten) +# enable_lastcall +# eval "incoming_${incoming_context}_${1}=\"$2\"" +# ;; + mailbox|allowtransfer|answerfirst|beforeanswer|timeout|menucontext|match|matchcaller|aftertimeout|aftertimeouttype) + eval "incoming_${incoming_context}_${1}=\"$2\"" + ;; + target|target_ITEM*) + append dialplan_incoming_${incoming_context} "$2" "&" + ;; + target_COUNT) ;; + *) logerror "Invalid option $1 in incoming" ;; + esac + } +} + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/dialplanconf.txt b/contrib/asterisk-xip/files/uci/dialplanconf.txt new file mode 100644 index 0000000000..eef930bd3f --- /dev/null +++ b/contrib/asterisk-xip/files/uci/dialplanconf.txt @@ -0,0 +1,84 @@ + +dialplangeneral + asterisk options: + static + writeprotect + canreinvite + clearglobalvars + + global settings for dialzone out: + lastdialed + lastdialedtype + + Global option for dialplanexten: + answerfirst + dialtimeout + allowtransfer - allow transfers in dialing + + voiceboxext Extension to use for voicebox set on local client + + international + internationalout + +dialplan {name} - Define a dialplan context + include + #=mailbox|voice|meetme|saytime|clock + +dialplanexten - Add number to Dialplan for ringing an extension + dialplan - Dialplan to add to + extension - Number to dial + type - Channel type sip/iax/local + target - Target + dialextension - Extension to dial + voicebox - Voicebox to fall back to + timeout - Timeout for dial + answerfirst - Answer before dialing + +dialplangoto + dialplan|extension|target) + +dialplansaytime + dialplan - Dialplan to add to + extension - Number to dial + zone - Time Zone to use + format - Format to use. + +dialzone {name} - Outgoing zone. + uses - Outgoing line to use: TYPE/Name + match (list) - Number to match + countrycode - The effective country code of this dialzone + international (list) - International prefix to match + localzone - dialzone for local numbers + addprefix - Prexix required to dial out. + localprefix - Prefix for a local call + +inzone + TODO + +incominggeneral + allowtransfer - Default Allow transfers for Dial() + timeout - Default timeout for incoming calls + answerfirst - Default value for incoming calls + mailbox - Default global mailbox for incoming calls + lastcall - Subroutine Context to store Last incoming call + lastcalltype - Method for calling lastcall (default goto) goto|gosub|macro|start + missed - Subroutine context to store last missed call dialplan context + missedtype - Method for calling 'missed' context goto|gosub|macro|start + +incoming {name} - Incoming zone + allowtransfer - Allow transfers for Dialed extension + timeout - Timeout for call + answerfirst - Answer the incoming call before Ringing + mailbox - Voicemail mailbox to use when + + beforeanswer - Time to ring before asterisk 'answers' and takes control of the call + menucontext - EXITCONTEXT for the ring once asterisk is handling the call + match - Dialed number to match + matchcaller - Caller to match + aftertimeout - Target macro/goto once the timeout has past, before voicemail + aftertimeouttype - Type of the target (macro|start|goto) + lastcall - Subroutine Context to store Last incoming call + lastcalltype - Method for calling lastcall (default goto) goto|gosub|macro|start + missed - Subroutine context to store last missed call dialplan context + missedtype - Method for calling 'missed' context goto|gosub|macro|start + diff --git a/contrib/asterisk-xip/files/uci/featureconf b/contrib/asterisk-xip/files/uci/featureconf new file mode 100755 index 0000000000..e336570ef5 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/featureconf @@ -0,0 +1,99 @@ +#!/bin/sh + +# Feature.conf +ast_add_conf feature +init_featureconf(){ + ast_add_reload feature + ast_enable_type feature + ast_enable_type featurepark + ast_enable_type featuremap + + feature_park_parkenabled=no + feature_park_parkext=700 + feature_park_parkpos="701-720" + feature_park_context=parkedcalls + feature_park_parkingtime=45 + feature_park_courtesytone=beep + feature_park_parkedplay=caller + feature_park_adsipark=yes + feature_park_findslot=first + feature_park_parkedmusicclass=default + feature_park_transferdigittimeout=3 + feature_park_xfersound=beep + feature_park_xferfailsound=beeperr + feature_park_pickupexten="*8" + feature_park_featuredigittimeout=500 + feature_park_atxfernoanswertimeout=15 +} + +feature_park_list="parkext parkpos context parkingtime \ +courtesytone parkedplay adsipark findslot parkedmusicclass \ +transferdigittimeout xfersound xferfailsound pickupexten \ +featuredigittimeout atxfernoanswertimeout" +feature_map_list="blindxfer disconnect automon atxfer parkcall" + +valid_features(){ + case $1 in + park) is_in_list $2 ${feature_park_list} parkenabled ; return $? ;; + map) is_in_list $2 ${feature_map_list} ; return $? ;; + *) return 1;; + esac +} + +create_featureconf(){ + file=${DEST_DIR}/features.conf + get_checksum feature_conf $file + + local isempty=1 + if [ $feature_park_parkenabled == no ] ; then + rm -f $file + isempty=2 + else + enable_module res_features + echo "${asteriskuci_gen}${N}[general]" > $file + for i in ${feature_park_list} ; do + eval value="\"\${feature_park_$i}\"" + [ ! -z "$value" ] && echo "$i=$value" >> $file + done + echo "${N}[featuremap]" >> $file + for i in ${feature_map_list} ; do + eval value="\"\${feature_map_$i}\"" + [ ! -z "$value" ] && echo "$i=$value" >> $file + done + fi + check_checksum "$feature_conf" "$file" || ast_feature_restart=$isempty + +} +handle_featurepark() { + handle_feature park +} +handle_featuremap() { + handle_feature map +} + +handle_feature() { + feature_type=$1 + option_cb() { + if valid_features ${feature_type} $1 $2 ; then + eval "feature_${feature_type}_$1=\"$2\"" + else + logerror "Invalid feature: $1" + fi + } +} + +append_dialplan_park(){ + local file=$1 + # Check for parked calls - add into available extensions + if [ ${feature_park_parkenabled} == yes ] && [ ! -z ${feature_park_context} ] ; then + add_dialplan_include extensions ${feature_park_context} + enable_module app_parkandannounce + enable_format gsm + fi +} + + +reload_feature() astcmd "module reload res_features.so" +unload_feature() astcmd "module unload res_features.so" + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/featureconf.txt b/contrib/asterisk-xip/files/uci/featureconf.txt new file mode 100644 index 0000000000..a47b223da6 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/featureconf.txt @@ -0,0 +1,25 @@ +See asterisk doco +featurepark + parkext + parkpos + context + parkingtime + courtesytone + parkedplay + adsipark + findslot + parkedmusicclass + transferdigittimeout + xfersound + xferfailsound + pickupexten + featuredigittimeout + atxfernoanswertimeout + +featuremap + blindxfer + disconnect + automon + atxfer + parkcall + diff --git a/contrib/asterisk-xip/files/uci/lastcall b/contrib/asterisk-xip/files/uci/lastcall new file mode 100755 index 0000000000..c5ec6e1c48 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/lastcall @@ -0,0 +1,119 @@ +#!/bin/sh +# +# Author: Michael Geddes <michael at frog dot wheelycreek dot net> +# Copyright 2008 Michael Geddes +# Licensed under GPL + +ast_add_module lastcall + +init_lastcall() { + # + ast_enable_type calllist + ast_enable_type calllistdefault +} + +check_add_calllist() { + local context=${opt_calllist_zonename} + if [ ! -z "$context" ] ; then + logdebug 1 "Adding calllist context ${context}" + [ -z $opt_calllist_dialplan ] && opt_calllist_dialplan=${opt_calllist_general_dialplan} + [ -z $opt_calllist_dialplan ] && opt_calllist_dialplan=extensions + [ -z $opt_calllist_listname ] && opt_calllist_listname=lastcall + [ -z $opt_calllist_length ] && opt_calllist_length=${opt_calllist_general_length} + [ -z $opt_calllist_length ] && opt_calllist_length=10 + [ -z $opt_calllist_dialcontext ] && opt_calllist_dialcontext=${opt_calllist_general_dialcontext} + [ -z $opt_calllist_dialcontext ] && opt_calllist_dialcontext=default + [ -z $opt_calllist_calltype ] && opt_calllist_calltype=macro + + [ -z ${opt_calllist_extension} ] \ + || add_dialplan_lastcall ${opt_calllist_dialplan} "${opt_calllist_extension}" "${opt_calllist_listname}" "${opt_calllist_length}" "${opt_calllist_tagname}" "${opt_calllist_dialcontext}" + enable_lastcall + add_section_lastcall ${context} "${opt_calllist_listname}" "${opt_calllist_length}" "${opt_calllist_calltype}" + fi + for i in zonename extension dialplan length listname ; do + eval "unset opt_calllist_${i}" + done +} + +add_section_lastcall() { + local context=$1 + local name=$2 + local queuelen=$3 + local calltype=$4 + + [ "${calltype}" = "macro" ] && context=macro-${1} + + if check_add_context ${context} ; then + local ext="exten => s," + case "${calltype}" in + gosub) + enable_module app_stack + append_dialplan_context ${context} "${ext}1,Macro(lastcallstore,\${EXTEN},${name},${queuelen})" + append_dialplan_context ${context} "${ext}n,Return" ;; + gosub_s) + enable_module app_stack + append_dialplan_context ${context} "${ext}1,Macro(lastcallstore,\${ARG1},${name},${queuelen})" + append_dialplan_context ${context} "${ext}n,Return" ;; + macro) + enable_module app_macro + append_dialplan_context ${context} "${ext}1,Macro(lastcallstore,\${ARG1},${name},${queuelen})" ;; + esac + else + logerror "Lastcall section ${context} already added" + fi +} + +handle_calllist() { + check_add calllist + logdebug 2 "Loading Call List: ${opt_calllist_zonename}" + opt_calllist_zonename=$1 + option_cb() { + case "$1" in + extension|dialplan|length|listname|calllist|tagname|dialcontext|calltype) + logdebug 1 "Setting opt_calllist_$1=\"${2}\"" + eval "opt_calllist_${1}=\"${2}\"" ;; + *) + logerror "Unknown option $1 in calllist ${opt_calllist_zonename}" ;; + esac + } +} + +handle_calllistdefault() { + logdebug 2 "Loading Call List General options" + option_cb() { + case $1 in + dialplan|length|dialcontext) + eval "opt_calllist_general_${1}=\"${2}\"" ;; + *) logerror "Unknown option $1 in calllistdefault" ;; + esac + } +} + +add_dialplan_lastcall(){ + local context=$1 + logdebug 1 "Adding Dialplan lastcall $1 $2" + check_add_context "$context" + enable_lastcall + local queue=$3 + local len=$4 + local tag=$5 + local dialcontext=$6 + [ "${queue}" == lastcall ] && queue= + append dialplan_context_${context} "exten => $2,1,Macro(lastcallapp,${queue},${len},${dialcontext},${tag})" "${N}" +} + +enable_lastcall() { + if [ "${dialplan_do_add_lastcall}" != "1" ] ; then + logdebug 2 "Enabling lastcall" + append dialplan_globals "LASTCALLZONE=\"${asterisk_zone}\"" "$N" + append_include "macros/lastcall.conf" + dialplan_do_add_lastcall=1 + enable_module app_macro + enable_module func_callerid + enable_module app_sayunixtime + enable_module app_playback + enable_module func_db + enable_format gsm + fi +} +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/lastcall.txt b/contrib/asterisk-xip/files/uci/lastcall.txt new file mode 100644 index 0000000000..cd86be3f1a --- /dev/null +++ b/contrib/asterisk-xip/files/uci/lastcall.txt @@ -0,0 +1,15 @@ + +calllistdefault + dialplan - Override default dialplan for calllist extensions to be added to. + length - Default Length of list + dialcontext - Default Context for dialing + +calllist {name} - Define a call list context + extension - Extension to review list + dialplan - Dialplan for extension to be added to (default to extensions) + length - Length of list + listname - Name of list to use + tagname - Sound file to use for name + dialcontext - Context to use when dialing + calltype - Type of context for the wrapper (required when calling it) macro|gosub + diff --git a/contrib/asterisk-xip/files/uci/meetmeconf b/contrib/asterisk-xip/files/uci/meetmeconf new file mode 100755 index 0000000000..d70016148a --- /dev/null +++ b/contrib/asterisk-xip/files/uci/meetmeconf @@ -0,0 +1,107 @@ +#!/bin/sh +# Meetme.conf (conference) + +ast_add_conf meetme +init_meetmeconf() { + ast_add_reload meetme + ast_enable_type meetmegeneral + ast_enable_type meetme + ast_enable_type dialplanmeetme +} +create_meetmeconf() { + file=${DEST_DIR}/meetme.conf + get_checksum meetme_conf $file + logdebug 1 "Creating meetme rooms: ${meetme_rooms}" + local isempty=1 + if [ -z ${meetme_rooms} ] ; then + rm -f $file + isempty=2 + else + echo "${asteriskuci_gen}${N}[general]" > $file + if [ ! -z ${meetme_audiobuffers} ] ; then + echo "audiobuffers=${meetme_audiobuffers}" >> $file + fi + echo "${N}[rooms]" >> $file + for i in ${meetme_rooms} ; do + for j in pin adminpin room ; do + eval meetme_$j=\${meetme_room_${i}_${j}} + done + if [ -z "${meetme_room}" ] ; then + meetme_room=$i + fi + local line="conf => ${meetme_room}" + if [ -z ${meetme_adminpin} ] ; then + if [ ! -z ${meetme_pin} ] ; then + line="${line},${meetme_pin}" + fi + else + line="${line},${meetme_pin},${meetme_adminpin}" + fi + echo "$line" >> $file + done + fi + check_checksum "$meetme_conf" "$file" || ast_meetme_restart=$isempty +} + +handle_meetmegeneral() { + option_cb() { + case $1 in + audiobuffers) + meetme_audiobuffers="$2" ;; + *) logerror "Invalid meetme general option $1" + esac + } +} + +handle_meetme() { + logdebug 2 "Add meetme room $1" + meetme_room="$1" + append meetme_rooms "$1" " " + enable_module app_meetme + option_cb() { + case $1 in + pin|adminpin|room) + logdebug 3 "Meetme option ${meetme_room}/${1}=$2" + eval meetme_room_${meetme_room}_${1}="$2" ;; + *) logerror "Invalid meetme option for $meetme_room : $1" + esac + } +} + +handle_dialplanmeetme() { + check_add dialplanmeetme + option_cb() { + case $1 in + dialplan|extension|room) + eval "dial_meetme_$1=\"$2\"" + esac + } +} + +check_add_dialplanmeetme() { + if [ ! -z "${dial_meetme_extension}" ] ; then + local ext="exten => ${dial_meetme_extension}," + + [ -z "${dial_meetme_dialplan}" ] && dial_meetme_dialplan=extensions + check_add_context ${dial_meetme_dialplan} + append dialplan_context_${dial_meetme_dialplan} "${ext}1,MeetMe(${dial_meetme_room})" "${N}" + append dialplan_context_${dial_meetme_dialplan} "${ext}n,HangUp" "${N}" + fi + for i in dialplan extension room ; do + eval "unset dial_meetme_$i" + done +} + +add_dialplan_meetme() { + local context=$1 + logdebug 1 "Adding Dialplan meetme $1 $2" + check_add_context "$context" + local ext="exten => $2," + append dialplan_context_${context} "${ext}1,MeetMe($3)" "${N}" + append dialplan_context_${context} "${ext}n,HangUp" "${N}" +} + +reload_meetme() astcmd "module reload app_meetme.so" +unload_meetme() astcmd "module unload app_meetme.so" + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/meetmeconf.txt b/contrib/asterisk-xip/files/uci/meetmeconf.txt new file mode 100644 index 0000000000..fc24e389c4 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/meetmeconf.txt @@ -0,0 +1,13 @@ + +meetmegeneral + audiobuffers + +meetme - add a meetme room + room - Number of room + pin - PIN for room + adminpin - PIN for room admin + +dialplanmeetme - add calling of meetme to a dialplan + dialplan - Dialplan to add to (default 'extensions') + extension - Extension to dial + room - Optional room to enter diff --git a/contrib/asterisk-xip/files/uci/moduleconf b/contrib/asterisk-xip/files/uci/moduleconf new file mode 100755 index 0000000000..d8ea6114a0 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/moduleconf @@ -0,0 +1,151 @@ +#!/bin/sh + +# Module.conf + +init_moduleconf() { + ast_add_reload module + ast_enable_type module + + for i in ${module_list} ; do + eval module_${i}=no + done + + enable_module app_dial + enable_module app_read + + enable_module app_verbose + enable_module pbx_config + enable_module pbx_functions + + enable_module app_transfer + module_chan_local=auto + # Sound files are all gsm + enable_format gsm + + enable_module app_func_strings +} + +# List of modules in sensible load order. +module_list="res_agi res_adsi res_config_mysql res_crypto res_smdi res_features \ + res_indications res_convert res_jabber res_monitor res_musiconhold res_speech \ + res_clioriginate pbx_ael pbx_config pbx_functions pbx_loopback pbx_realtime \ + pbx_spool pbx_wilcalu func_base64 func_callerid func_cdr func_channel func_cut \ + func_db func_enum func_env func_global func_groupcount func_language func_logic \ + func_moh func_rand func_realtime func_sha1 func_strings func_timeout func_uri \ + cdr_csv cdr_custom cdr_manager cdr_mysql cdr_pgsql cdr_sqlite chan_agent \ + chan_alsa chan_gtalk chan_h323 chan_iax2 chan_local chan_sip format_au \ + format_g723 format_g726 format_g729 format_gsm format_h263 format_h264 \ + format_ilbc format_jpeg format_mp3 format_pcm format_pcm_alaw format_sln \ + format_vox format_wav format_wav_gsm app_alarmreceiver app_amd app_authenticate \ + app_cdr app_chanisavail app_channelredirect app_chanspy app_controlplayback \ + app_cut app_db app_dial app_dictate app_directed_pickup app_directory app_disa \ + app_dumpchan app_echo app_enumlookup app_eval app_exec app_externalivr \ + app_followme app_forkcdr app_getcpeid app_groupcount app_hasnewvoicemail \ + app_ices app_image app_lookupblacklist app_lookupcidname app_macro app_math \ + app_md5 app_meetme app_milliwatt app_mixmonitor app_morsecode \ + app_parkandannounce app_playback app_privacy app_queue app_random app_read \ + app_readfile app_realtime app_record app_sayunixtime app_senddtmf app_sendtext \ + app_setcallerid app_setcdruserfield app_setcidname app_setcidnum app_setrdnis \ + app_settransfercapability app_sms app_softhangup app_speech_utils app_stack \ + app_system app_talkdetect app_test app_transfer app_txtcidname app_url \ + app_userevent app_verbose app_voicemail app_waitforring app_waitforsilence \ + app_while codec_a_mu codec_adpcm codec_alaw codec_g726 codec_gsm codec_ilbc \ + codec_lpc10 codec_speex codec_ulaw" + +# Enable a module - for use by other scripts +enable_module() { + logdebug 3 "Enable ${1}" + eval module_${1}=yes +} + +module_enabled() { + eval local is_enabled="\${module_${1}}" + if [ ${is_enabled} == "no" ] ; then + return 1 + else + return 0 + fi +} + +# Enable a sound format - for use by other scripts +enable_format() { + while [ ! -z $1 ] ; do + case $1 in + gsm) + enable_module format_gsm + enable_module codec_gsm + [ "${module_format_wav}" = "yes" ] && enable_module format_wav_gsm ;; + wav) + enable_module format_wav + [ "${module_format_gsm}" = "yes" ] && enable_module format_wav_gsm ;; + alaw) + enable_module codec_adpcm + enable_module codec_alaw + [ "${module_format_pcm}" = "yes" ] && enable_module format_pcm_alaw ;; + pcm) + enable_module format_pcm_alaw + [ "${module_format_alaw}" = "yes" ] && enable_module format_pcm_alaw ;; + ulaw) + enable_module codec_ulaw ;; + g729|g726|g723) + enable_module format_g726 + enable_module codec_g726 ;; + ilbc) + enable_module format_ilbc + enable_module codec_ilbc ;; + sln) enable_module format_sln ;; + mp3) enable_module format_mp3 ;; + vox) enable_module format_vox ;; + speex) enable_module codec_speex ;; + esac + shift + done +} + +create_moduleconf() { + local file=${DEST_DIR}/modules.conf + + get_checksum module_conf $file + + rm -f ${file}.orig + [ -f "${file}" ] && mv ${file} ${file}.orig + + echo "${asteriskuci_gen}[modules]${N}autoload=yes" > $file + for i in ${module_list} ; do + eval res=\${module_${i}} + case $res in + yes) echo "load => $i.so" >> $file ;; + no) echo "noload => $i.so" >> $file ;; + esac + done + echo "${N}[global]${N}chan_modem.so=no" >> $file + + check_checksum "$module_conf" "$file" || ast_module_restart=1 +} + +reload_module() { + local file=${DEST_DIR}/modules.conf + local cmd=`diff ${file}.orig ${file} -u -U0 | grep '^+\(no\)\?load' | sed 's/+load[[:space:]]*=>[[:space:]]*\(.*\)$/\"module load \1\"/' | sed 's/+noload[[:space:]]*=>[[:space:]]*\(.*\)$/\"module unload \1\"/'| tr '\n' ' '` + [ "${testing_mode}" != "1" ] && rm -f ${file}.orig + logdebug 3 "Module reload: ${N}$cmd" + eval "astcmds $cmd" +} + + +valid_module() { + is_in_list $1 ${module_list} + return $? +} + +handle_module() { + option_cb() { + if valid_module $1 ; then + eval module_$1="$2" + else + logerror "Invalid module: $1" + fi + } +} + + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/moduleconf.txt b/contrib/asterisk-xip/files/uci/moduleconf.txt new file mode 100644 index 0000000000..7b083b6db0 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/moduleconf.txt @@ -0,0 +1,151 @@ + +module - Enable modules (yes/no/auto) + res_agi + res_adsi + res_config_mysql + res_crypto + res_smdi + res_features + res_indications + res_convert + res_jabber + res_monitor + res_musiconhold + res_speech + res_clioriginate + pbx_ael + pbx_config + pbx_functions + pbx_loopback + pbx_realtime + pbx_spool + pbx_wilcalu + func_base64 + func_callerid + func_cdr + func_channel + func_cut + func_db + func_enum + func_env + func_global + func_groupcount + func_language + func_logic + func_moh + func_rand + func_realtime + func_sha1 + func_strings + func_timeout + func_uri + cdr_csv + cdr_custom + cdr_manager + cdr_mysql + cdr_pgsql + cdr_sqlite + chan_agent + chan_alsa + chan_gtalk + chan_h323 + chan_iax2 + chan_local + chan_sip + format_au + format_g723 + format_g726 + format_g729 + format_gsm + format_h263 + format_h264 + format_ilbc + format_jpeg + format_mp3 + format_pcm + format_pcm_alaw + format_sln + format_vox + format_wav + format_wav_gsm + app_alarmreceiver + app_amd + app_authenticate + app_cdr + app_chanisavail + app_channelredirect + app_chanspy + app_controlplayback + app_cut + app_db + app_dial + app_dictate + app_directed_pickup + app_directory + app_disa + app_dumpchan + app_echo + app_enumlookup + app_eval + app_exec + app_externalivr + app_followme + app_forkcdr + app_getcpeid + app_groupcount + app_hasnewvoicemail + app_ices + app_image + app_lookupblacklist + app_lookupcidname + app_macro + app_math + app_md5 + app_meetme + app_milliwatt + app_mixmonitor + app_morsecode + app_parkandannounce + app_playback + app_privacy + app_queue + app_random + app_read + app_readfile + app_realtime + app_record + app_sayunixtime + app_senddtmf + app_sendtext + app_setcallerid + app_setcdruserfield + app_setcidname + app_setcidnum + app_setrdnis + app_settransfercapability + app_sms + app_softhangup + app_speech_utils + app_stack + app_system + app_talkdetect + app_test + app_transfer + app_txtcidname + app_url + app_userevent + app_verbose + app_voicemail + app_waitforring + app_waitforsilence + app_while + codec_a_mu + codec_adpcm + codec_alaw + codec_g726 + codec_gsm + codec_ilbc + codec_lpc10 + codec_speex + codec_ulaw + diff --git a/contrib/asterisk-xip/files/uci/mohconf b/contrib/asterisk-xip/files/uci/mohconf new file mode 100755 index 0000000000..9963108cf0 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/mohconf @@ -0,0 +1,74 @@ +#!/bin/sh + +# Music on Hold + +ast_add_conf moh +init_mohconf() { + ast_add_reload moh + ast_enable_type moh + ast_enable_type musiconhold +} + + +handle_musiconhold() handle_moh "$1" + +moh_list="name" +moh_optlist="mode directory random application format" + +valid_moh() { + is_in_list $1 ${moh_list} ${moh_optlist} + return $? +} + +handle_moh() { + check_add moh + moh_context=$1 + logdebug 1 "Loading MOH context: ${moh_context}" + + enable_module res_musiconhold + + option_cb() { + if valid_moh $1 $2 ; then + eval "moh_var_${1}=\"$2\"" + else + logerror "Invalid music-on-hold option for ${moh_context} : $1" + fi + } +} + +check_add_moh() { + if [ ! -z "${moh_var_directory}" ] ; then + [ -z "${moh_var_name}" ] && moh_var_name=default + [ -z "${moh_var_mode}" ] && moh_var_mode=files + append moh_lines "[${moh_var_name}]" "${N}${N}" + + for i in ${moh_optlist} ; do + eval "local curopt=\"\${moh_var_$i}\"" + [ -z "${curopt}" ] || append moh_lines "$i=${curopt}" "${N}" + done + fi + for i in ${moh_list} ${moh_optlist} ; do + eval "unset moh_var_$i" + done +} + +create_mohconf() { + file=${DEST_DIR}/musiconhold.conf + get_checksum moh_conf $file + local isempty=1 + if [ -z "${moh_lines}" ] ; then + isempty=2 + rm -f $file + else + echo "${asteriskuci_gen}" > $file + echo "${moh_lines}" >> $file + unset moh_lines + fi + check_checksum "$moh_conf" "$file" || ast_moh_restart=$isempty +} + +reload_moh() astcmd "moh reload" +unload_moh() astcmd "module unload res_musiconhold.so" + + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/mohconf.txt b/contrib/asterisk-xip/files/uci/mohconf.txt new file mode 100644 index 0000000000..232174062b --- /dev/null +++ b/contrib/asterisk-xip/files/uci/mohconf.txt @@ -0,0 +1,8 @@ + +musiconhold/moh - Musicon-on-hold context + name - MOH context name + mode - Playback mode (files) + directory - Directory of music + random - Playback is random + application - Application to use + format - file format of files diff --git a/contrib/asterisk-xip/files/uci/sipiaxconf b/contrib/asterisk-xip/files/uci/sipiaxconf new file mode 100755 index 0000000000..f3f0720207 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/sipiaxconf @@ -0,0 +1,545 @@ +#!/bin/sh +# Sip / IAX extensions + +add_incoming_context() { + local context=$1 + eval "local added=\${dialplan_incoming_${context}_added}" + if [ "${added}" != "1" ] ; then + append dialplan_extensions_incoming "${context}" " " + eval "dialplan_incoming_${context}_added=1" + fi + +} + +# Add to incoming ringing +add_incoming() { + local rhs="$3" + + while [ ! -z "$rhs" ] ; do + cur=${rhs%%,*} + nvar=${rhs#*,} + add_incoming_context ${cur} + append dialplan_incoming_${cur} "$1/$2" "&" + [ "$nvar" == "$rhs" ] && break + rhs=${nvar} + done +} + +# Add to internal extensions +add_extension() { + logdebug 1 "Adding $1/$2 extension to $3" + (eval [ -z "\${dialplan_ext_$2}" ] )\ + && append dialplan_exts "$3" " " + local lower=`echo $1|tr [A-Z] [a-z]` + eval "${lower}_last_extension=\"$3\"" + append dialplan_ext_$3 $1/${2} "&" +} + +check_append_local() { + local extension="${1}" + logdebug 3 "added local context for ${1}" + eval "local isadded=\"\${dialplan_add_local_${extension}-0}\"" + if [ "$isadded" != "1" ] ; then + eval "dialplan_add_local_${extension}=1" + append dialplan_locals "$extension" + eval "dialplan_local_${1}_context=\"${2}\"" + eval "dialplan_local_${1}_selfmailbox=\"${3}\"" + eval "dialplan_local_${1}_mailbox=\"${4}\"" + return 0 + else + return 1 + fi +} +append_dialplan_locals(){ + for i in ${dialplan_locals} ; do + local extension=$i + for x in context selfmailbox mailbox ; do + eval "x_${x}=\${dialplan_local_${i}_${x}}" + done + local newcontext=local_${extension} + + if check_add_context ${newcontext} ; then + # add_dialplan_voice ${newcontext} ${x_last_extension} ${x_last_mailbox} + # Make sure as much is matched as possible + #add_dialplan_goto ${newcontext} _[0-9#*+]. ${x_last_context} + # add_dialplan_include ${newcontext} ${x_last_context} + + append_dialplan_context ${newcontext} "exten => _.,1,Set(CALLERID(num)=${extension})" + if [ ! -z "${x_mailbox}" ] ; then + [ "${x_selfmailbox}" = "yes" ] && append_dialplan_context ${newcontext} "exten => ${extension},2,VoiceMailMain(${x_mailbox})" + [ ! -z "${dialplan_voiceboxext}" ] && append_dialplan_context ${newcontext} "exten => ${dialplan_voiceboxext},2,VoiceMailMain(${x_mailbox})" + fi + append_dialplan_context ${newcontext} "exten => _.,2,Goto(${x_context},\${EXTEN},1)" + fi + done +} + +# Sip + +check_add_sipitems() { + if [ "${sip_doregister}" == "1" ] ; then + local line="register => ${sip_last_username}@${sip_last_fromdomain}:${sip_last_secret}:${sip_last_username}@${sip_sectionname}" + case ${sip_last_registerextension} in + -) line="$line/${sip_last_username}" ;; + .*) line="$line/${sip_last_registerextension}" ;; + esac + append sip_register "$line" "$N" + sip_doregister=0 + fi + do_check_add_items sip +} +check_add_iaxitems() { + do_check_add_items iax +} + +do_check_add_items(){ + + for i in type last_host last_context selfmailbox last_extension last_mailbox ; do + eval "x_${i}=\"\${${1}_${i}-}\"" + done + + if [ ! -z "${x_last_context}" ] ; then + if [ ! -z "${x_last_extension}" ] ; then + [ "${x_last_context}" = "-" ] && eval "x_last_context=\"\${${1}_opt_context}\"" + check_append_local "${x_last_extension}" "${x_last_context}" "${x_selfmailbox}" "${x_last_mailbox}" + x_last_context=local_${x_last_extension} + fi + if [ "${x_last_context}" != "-" ] ; then + append ${1}_sections "context=${x_last_context}" "$N" + fi + if [ "${x_type}" != "user" -a -z "${x_last_host}" ] ; then + append ${1}_sections "host=dynamic" "$N" + fi + fi + + for i in last_username last_fromdomain last_secret last_username \ + sectionname last_fromuser last_context last_extension last_mailbox last_type last_host ; do + eval unset $1_$i + done + + eval ${1}_selfmailbox=no + eval ${1}_last_registerextension=- +} + +reload_sip() { + astcmd "sip reload" + return 1 # reboot +} +unload_sip() astcmd "unload chan_sip.so" + +rtp_option_list="rtpstart rtpend rtpdtmftimeout rtcpinterval rtpchecksums" +# Validate RTP options +valid_rtp_option() { + is_in_list $1 ${rtp_option_list} +} + +# Validate sip options, depending on context. +valid_sipiax_option() { + local use_glob=1 + local use_glob_iax=1 + local use_glob_sip=1 + local use_user=1 + local use_peer=1 + local use_user_sip=1 + local use_user_iax=1 + local use_peer_sip=1 + local use_peer_iax=1 + case "$1" in + globalsip) + use_glob_sip=0 + use_glob=0 ;; + usersip) + use_glob_sip=0 + use_glob=0 + use_user=0 ;; + peersip|friendsip) + use_glob_sip=0 + use_glob=0 + use_user=0 + use_peer=0 + use_user_sip=0 + use_peer_sip=0 ;; + globaliax) + use_glob_iax=0 + use_glob=0 ;; + useriax) + use_glob_iax=0 + use_glob=0 + use_user=0 ;; + peeriax|friendiax) + use_glob_iax=0 + use_glob=0 + use_user=0 + use_peer=0 + use_user_iax=0 + use_peer_iax=0 ;; + esac + + case "$2" in + writeprotect|static) return ${use_glob_iax} ;; +# Integer + port|\ + maxexpirey|\ + rtptimeout|\ + rtpholdtimeout|\ + defaultexpirey|\ + registertimeout|\ + registerattempts|\ + call-limit) return ${use_glob_sip} ;; +# ip addr + bindaddr|\ + externip) return ${use_glob_sip} ;; +# net/mask + localnet) return ${use_glob_sip} ;; + permit|\ + deny) return ${use_user_sip} ;; +# Domain name + realm|\ + domain) return ${use_glob_sip} ;; +# valid context + context) return ${use_glob} ;; +# Mime type + notifymimetype) return ${use_glob_sip} ;; +# Yes/No + canreinvite) return ${use_glob} ;; + nat|allowoverlap|allowsubscribe|allowtransfer|\ + videosupport) return ${use_glob_sip} ;; + pedantic|\ + trustrpid|\ + promiscredir|\ + useclientcode) return ${use_user_sip} ;; +# Enums + dtmfmode) return ${use_glob_sip} ;; + type) return ${use_user} ;; + insecure|callingpres|\ + progressinband) return ${use_user_sip} ;; +# List + allow|\ + disallow) return ${use_glob_sip} ;; +# Register string + register) return ${use_glob_sip} ;; +# String + username|secret|md5secret|host|\ + mailbox) return ${use_user} ;; + auth) return ${use_user_iax} ;; + callgroup|pickupgroup|language|accountcode|\ + setvar|callerid|amaflags|subscribecontext|\ + maxcallbitrate|rfc2833compensate|\ + mailbox) return ${use_user_sip};; + template|fromdomain|regexten|fromuser|\ + qualify|defaultip|sendrpid|\ + outboundproxy) return ${use_peer_sip};; + extension) return 0;; + *) return 1;; + esac +} + +ast_add_conf sip +init_sipconf() { + ast_add_reload sip + ast_enable_type sipgeneral + ast_enable_type sip + ast_enable_type target + + sip_opt_port=5060 + sip_opt_bindaddr=0.0.0.0 + sip_opt_context=default + sip_opt_maxexpirey=3600 + sip_opt_defaultexpirey=3600 + sip_opt_notifymimetype=text/plain + sip_opt_rtptimeout=60 + sip_opt_rtpholdtimeout=300 + config_get WAN_IP wan ipaddr + # TODO check why the above does not work all the time + if [ -z "${WAN_IP}" ] ; then + config_get WAN_IF wan ifname + WAN_IP=$(ifconfig ${WAN_IF} | grep "inet addr:" | sed 's/^.*inet addr:\([^ ]*\) .*$/\1/') + fi + + sip_opt_externip=${WAN_IP} + + sip_opt_realm=asterisk + config_get LAN_MASK lan netmask + config_get LAN_IP lan ipaddr + LAN_NET=$(/bin/ipcalc.sh $LAN_IP $LAN_MASK | grep NETWORK | cut -d= -f2) + sip_opt_localnet=$LAN_NET/$LAN_MASK + + # default to ulaw only + sip_opt_allow= + sip_opt_registertimeout=20 + sip_opt_registerattempts=10 + sip_opt_canreinvite=no + + sip_sections= +} + +sip_list="port bindaddr context maxexpirey defaultexpirey notifymimetype \ +rtptimeout rtpholdtimeout realm domain localnet externip" + +create_sipconf() { + + append_dialplan_locals + + file=${DEST_DIR}/sip.conf + get_checksum sip_conf $file + local isempty=1 + if [ -z "${sip_sections}" ] ; then + rm -f $file + isempty=2 + else + [ -z "${sip_opt_domain}" ] && sip_opt_domain=${sip_opt_realm} + + echo "${asteriskuci_gen}[general]" > $file + for i in ${sip_list} ; do + eval value=\$sip_opt_$i + [ ! -z "$value" ] && ( echo "$i=$value" >> $file ) + done + echo "disallow=all" >> $file + local rhs="${sip_opt_allow}" + if [ -z "$rhs" ] ; then + rhs=ulaw + fi + while [ ! -z "$rhs" ] ; do + cur=${rhs%%,*} + nvar=${rhs#*,} + enable_format ${cur} + echo "allow=${cur}" >> $file + [ "$nvar" == "$rhs" ] && break + rhs=${nvar} + done + + echo "${N}${sip_register}${N}${N}${sip_sections}" >> $file + unset sip_register + unset sip_sections + fi + check_checksum "$sip_conf" "$file" || ast_sip_restart=$isempty +} + + +handle_sipgeneral() { + option_cb(){ + if valid_sipiax_option globalsip $1 $2 ; then + case "$1" in + host) + if [ -z "$2" ] ; then + sip_opt_host=dynamic + else + sip_opt_host="$2" + fi ;; + allow_LENGTH) ;; + allow|allow_ITEM*) + append sip_opt_allow "$2" "," ;; + *) eval "sip_opt_$1=\"\$2\"" ;; + esac + elif valid_rtp_option $1 $2 ; then + eval "rtp_opt_$1=\"\$2\"" + else + logerror "Invalid SIP global option: $1" + fi + } +} + +handle_sip() { + check_add sipitems + append sip_sections [$1] "$N$N" + enable_module chan_sip + sip_sectionname=${1#sip_} + sip_type=peer + sip_doregister=0 + sip_last_context=- + sip_last_doregister=- + sip_selfmailbox=no + option_cb() { + logdebug 3 "SIP/${sip_sectionname}: '$1' '$2'" + case $1 in + type) sip_type=$2 + append sip_sections "$1=$2" "$N" + ;; + register) + if [ "$2" == "yes" ]; then + sip_doregister=1 + fi ;; + registerextension) eval sip_last_$1="$2";; + allow|allow_ITEM*) split_append sip_sections allow= "$2" "${N}" enable_format ;; + extension|extension_ITEM*) add_extension SIP ${sip_sectionname} "$2" ;; + context) sip_last_context="$2" ;; + selfmailbox) sip_selfmailbox="$2" ;; + incoming|incoming_ITEM*) + add_incoming SIP ${sip_sectionname} "$2" ;; + timeout|prefix|internationalprefix|alwaysinternational|countrycode) + eval "target_$1_SIP_${sectionname}=\"$2\"" + ;; + allow_LENGTH|incoming_LENGTH|extension_LENGTH) ;; + *) + eval sip_last_$1="$2" + if valid_sipiax_option ${sip_type}sip $1 $2 ; then + append sip_sections "$1=$2" "$N" + else + logerror "Invalid SIP option for ${sip_type}: $1" + fi + esac + } +} + +# rtp.conf + +ast_add_conf rtp +init_rtpconf() { + ast_add_reload rtp + rtp_opt_rtpstart=5000 + rtp_opt_rtpend=31000 + rtp_opt_rtpchecksums= + rtp_opt_rtpdtmftimeout= + rtp_opt_rtcpinterval=5000 +} + +create_rtpconf() { + file=${DEST_DIR}/rtp.conf + get_checksum rtp_conf $file + local isempty=1 + if module_enabled chan_sip ; then + echo "${asteriskuci_gen}[general]" > $file + for i in $rtp_option_list ; do + eval "local val=\"\$rtp_opt_$i\"" + if [ ! -z "$val" ] ; then + lhs=$i + case "$i" in + rtpdtmftimeout) lhs=dtmftimeout + esac + echo "$lhs=$val" >> $file + fi + done + else + rm -f $file + isempty=2 + fi + + check_checksum "$rtp_conf" "$file" || ast_rtp_restart=$isempty +} +reload_rtp() astcmd "rtp reload" +unload_rtp() astcmd "unload rtp" + + +# Iax + +ast_add_conf iax + +init_iaxconf() { + ast_add_reload iax + ast_enable_type iaxgeneral + ast_enable_type iax + + return 0 +} + +create_iaxconf() { + local file=$DEST_DIR/iax.conf + get_checksum iax_conf $file + local isempty=1 + if [ -z "${iax_sections}" ] ; then + rm -f $file + isempty=2 + else + echo "${asteriskuci_gen}${iax_general}$N$N${iax_sections}" > $file + fi + check_checksum "$iax_conf" "$file" || ast_iax_restart=${isempty} +} + +handle_iaxgeneral() { + iax_general="[general]" + option_cb() { + case $1 in + allow_LENGTH) ;; + allow|allow_ITEM*) split_append iax_general allow= "$2" "${N}" enable_format ;; + *) + if valid_sipiax_option globaliax $1 $2 ; then + eval "iax_opt_$1=\"$2\"" + append iax_general "$1=$2" "$N" + else + logerror "Invalid IAX global option: $1" + fi ;; + esac + } +} + +handle_iax() { + check_add iaxitems + append iax_sections "[$1]" "$N$N" + iax_type=peer + iax_sectionname="${1#iax_}" + iax_last_context=- + iax_selfmailbox=no + enable_module chan_iax2 + option_cb() { + case $1 in + type) + iax_type=$2 + append iax_sections "type=$2" "$N" ;; + allow_LENGTH) ;; + allow|allow_ITEM*) + split_append iax_sections allow= "$2" "${N}" enable_format ;; + extension) + logdebug 1 "Adding IAX extension $2 for $iax_sectionname" + eval [ -z "\${dialplan_ext_$2}" ] && dialplan_exts="${dialplan_exts} $2" + iax_last_extension="$2" + append dialplan_ext_$2 "IAX2/${iax_sectionname}" "&" ;; + extension) add_extension IAX ${iax_sectionname} "$2" ;; + context) + eval iax_last_context="$2" ;; + selfmailbox) + eval iax_selfmailbox="$2" ;; + incoming) + add_incoming IAX ${iax_sectionname} "$3" ;; + timeout|prefix|internationalprefix|alwaysinternational|countrycode) + eval "target_$1_IAX_${sectionname}=\"$2\"" ;; + *) + eval iax_last_$1="$2" + if valid_sipiax_option ${iax_type}iax $1 $2 ; then + append iax_sections "$1=$2" "$N" + else + logerror "Invalid IAX option for ${iax_type}: $1" + fi + esac + } +} + +reload_iax() { + astcmd "iax2 reload" + return 1 +} +unload_iax() astcmd "unload chan_iax2.so" + +handle_target() { + # Target name + targettype=${1%[-_]*} + if [ ${targettype} == $1 ] ; then + logerror "No target type specified (SIP-$1 IAX-$1)" + return 1 + fi + targetname=${1#*[-_]} + + case $targettype in + [Ss][Ii][Pp]) handle_dialtarget SIP $targetname ;; + [Ii][Aa][Xx]) handle_dialtarget IAX $targetname ;; + *) logerror "Invalid target type specified: $targettype" + esac +} + +# Set up options sip/iax targets for outgoing sip/iax +handle_dialtarget() { + # Dialzone target option + areatype=$1 + areaname=$2 + logdebug 1 "Dialzone Target for ${areatype}/${areaname}" + option_cb(){ + case $1 in + timeout|prefix|internationalprefix|alwaysinternational|countrycode) + eval target_$1_${areatype}_${areaname}=$2 + ;; + *) + logerror "Invalid target for $areatype/$areaname: ${1}" + esac + } +} + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/sipiaxconf.txt b/contrib/asterisk-xip/files/uci/sipiaxconf.txt new file mode 100644 index 0000000000..ce2da498c7 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/sipiaxconf.txt @@ -0,0 +1,114 @@ + +target (SIP|IAX)_{name} - Handle options for outgoing dialing + timeout - Timeout for dialing out with this target + prefix - Prefix required to dial out on this target + internationalprefix- Prefix required to dial internation on this target + alwaysinternational- True if this target always requires internation prefix + countrycode - Default International country code for this target + +sipgeneral +sip [sip_]{name} + type - Type of sip connection - very important. + register - Set to yes to register + registerextension - 'Extension' to use in register string in place of username + + + extension - Extension to use for this phone (doesn't have to be unique) + selfmailbox - Set to yes to have dialing own extensions go to mailbox + incoming (list) - Specify the incoming context for ringing + + timeout - Timeout for dialing out with this target + prefix - Prefix required to dial out on this target + internationalprefix- Prefix required to dial internation on this target + alwaysinternational- True if this target always requires internation prefix + countrycode - Default International country code for this target + + rtpstart - rtp.conf option + rtpend - rtp.conf option + rtpdtmftimeout - rtp.conf option dtmftimeout + rtcpinterval - rtp.conf option + rtpchecksums - rtp.conf option + + +iaxgeneral +iax [iax_]{name} + extension Extensions to use for this phone + + selfmailbox - Set to yes to have dialing own extensions go to mailbox + incoming (list) - Specify the incoming context for ringing + + timeout - Timeout for dialing out with this target + prefix - Prefix required to dial out on this target + internationalprefix- Prefix required to dial internation on this target + alwaysinternational- True if this target always requires internation prefix + countrycode - Default International country code for this target + +General asterisk options - see asterisk doco + +Options |Type |sip |sip |iax |iax + | |general| |general| +-----------------+--------+-------+-----+-------+----- +writeprotect Integer no no yes yes +static Integer no no yes yes +port Integer yes yes no no +maxexpirey Integer yes yes no no +rtptimeout Integer yes yes no no +rtpholdtimeout Integer yes yes no no +defaultexpirey Integer yes yes no no +registertimeout Integer yes yes no no +registerattempts Integer yes yes no no +call-limit Integer yes yes no no +# ip addr +bindaddr IP Addr yes yes no no +externip IP Addr yes yes no no +localnet Net/mask yes yes no no +permit Net/mask no yes no no +deny Net/mask no yes no no +realm Domain yes yes no no +domain Domain yes yes no no +context context yes yes yes yes +notifymimetype Mimetype yes yes no no +canreinvite Yes/No yes yes yes yes +nat Yes/No yes yes no no +allowoverlap Yes/No yes yes no no +allowsubscribe Yes/No yes yes no no +allowtransfer Yes/No yes yes no no +videosupport Yes/No yes yes no no +pedantic Yes/No no yes no no +trustrpid, Yes/No no yes no no +promiscredir Yes/No no yes no no +useclientcode Yes/No no yes no no +dtmfmode Enum yes yes no no +type Enum no yes no yes +insecure Enum no yes no no +callingpres Enum no yes no no +progressinband Enum no yes no no +allow List yes yes yes yes +disallow List yes yes yes yes +register Register yes yes no no +username String no yes no yes +secret String no yes no yes +md5secret String no yes no yes +host String no yes no yes +mailbox String no yes no yes +auth String no no no yes +callgroup String no yes no no +pickupgroup String no yes no no +language String no yes no no +accountcode String no yes no no +setvar String no yes no no +callerid String no yes no no +amaflags String no yes no no +subscribecontext String no yes no no +maxcallbitrate String no yes no no +rfc2833compensate String no yes no no +mailbox String no yes no no +template String no peer no no +fromdomain String no peer no no +regexten String no peer no no +fromuser String no peer no no +qualify String no peer no no +defaultip String no peer no no +sendrpid String no peer no no +outboundproxy String no peer no no + diff --git a/contrib/asterisk-xip/files/uci/talkclock b/contrib/asterisk-xip/files/uci/talkclock new file mode 100755 index 0000000000..20e36065f8 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/talkclock @@ -0,0 +1,48 @@ +#!/bin/sh + +ast_add_module clock + +init_clock() { + ast_enable_type dialplanclock +} + +add_dialplan_talkclock() { + local context=$1 + local zone=${asterisk_zone} + [ ! -z "$3" ] && zone="$3" + local date_format="$4" + local time_format="$5" + logdebug 1 "Adding Dialplan talking clock $1 $2" + check_add_context "$context" + local ext="exten => $2," + if [ "${dialplan_add_include_clock}" != 1 ] ; then + dialplan_add_include_clock=1 + enable_format gsm + enable_module app_sayunixtime + append_include "macros/clock.conf" + fi + append dialplan_context_${context} "${ext}1,Macro(talkingclock,${time_format},${date_format},${zone})" "${N}" +} + +handle_dialplanclock() { + check_add dialplanclock + option_cb() { + case $1 in + dialplan|extension|zone|timeformat|dateformat) + eval "dial_clock_$1=\"$2\"" ;; + esac + } +} + +check_add_dialplanclock() { + if [ ! -z "${dial_clock_extension}" ] ; then + [ -z ${dial_clock_dialplan} ] && dial_clock_dialplan=default + add_dialplan_talkclock "${dial_clock_dialplan}" "${dial_clock_extension}" \ + "${dial_clock_zone}" "${dial_clock_dateformat}" "${dial_clock_timeformat}" + fi + for i in dialplan extension zone timeformat dateformat ; do + eval "unset dial_clock_$i" + done +} + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/files/uci/talkclock.txt b/contrib/asterisk-xip/files/uci/talkclock.txt new file mode 100644 index 0000000000..9d4cdf3608 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/talkclock.txt @@ -0,0 +1,7 @@ + +dialplanclock + dialplan - dialplan to add clock to + extension - extensions for talking clock + zone - Timezone to use + timeformat - Time Format to use + dateformat - Date format to use diff --git a/contrib/asterisk-xip/files/uci/voicemailconf b/contrib/asterisk-xip/files/uci/voicemailconf new file mode 100755 index 0000000000..83151f0b99 --- /dev/null +++ b/contrib/asterisk-xip/files/uci/voicemailconf @@ -0,0 +1,217 @@ +#!/bin/sh + +# Voicemail.conf + +ast_add_conf voicemail + +init_voicemailconf() { + ast_add_reload voicemail + + ast_enable_type voicegeneral + ast_enable_type voicemail + ast_enable_type voicezone + ast_enable_type dialplanvoice + + voice_format="wav49|gsm|wav" + voice_serveremail= + voice_attach=no + voice_skipms=3000 + voice_maxsilence=10 + voice_silencethreshold=128 + voice_maxlogins=3 + voice_emaildateformat="%A, %B %d, %Y at %r" + voice_sendvoicemail=no + voice_maxmsg=100 + voice_maxmessage=180 + voice_minmessage=3 + voice_maxgreet=60 + return 0 +} + +voicegeneral_list="format serveremail attach skipms maxsilence silencethreshold maxlogins emaildateformat sendvoicemail maxmsg maxmessage minmessage maxgreet" +voicegeneral_ext_list="" + +valid_voicemail(){ + is_in_list $1 ${voicegeneral_list} ${voicegeneral_ext_list} + return $? +} + +voicebox_list="context number password name email pager" + +voicebox_listopt="tz attach serveremail saycid dialout callback review operator envelope sayduration saydurationm" + +valid_voicebox() { + is_in_list $1 ${voicebox_list} ${voicebox_listopt} + return $? +} + +check_add_voicebox() { + if [ ! -z ${voicebox_number} ] ; then + [ -z "${voicebox_context}" ] && voicebox_context=default + logdebug 1 "Adding Voicebox ${voicebox_number} in ${voicebox_context}" + # Construct the voicebox line + local line="$voicebox_number => " + [ -z ${voicebox_tz} ] && voicebox_tz=homeloc + [ -z ${voicebox_name} ] && voicebox_name=OpenWRT + + for i in password name email pager ; do + eval "local value=\"\${voicebox_$i}\"" + line="${line}${value}," + done + + # Then add named options. + for i in ${voicebox_listopt} ; do + eval val=\${voicebox_$i} + [ -z ${val} ] || append line "$i=$val" \| + done + + # Check if the current voicebox context has anything + eval local cur=\${voicebox_section_$voicebox_context} + # if not add it to the list of contexts used + [ -z $cur ] && append voice_contextlist "${voicebox_context}" " " + + # Then add the voicebox line to the context + logdebug 4 "Add Voicebox $line to ${voicebox_context}" + append voicebox_section_${voicebox_context} "$line" "$N" + fi + + # Then clear the settings for the next one. + for i in ${voicebox_list} ${voicebox_listopt} ; do + eval unset voicebox_$i + done +} + +create_voicemailconf() { + # Construct the file + file=${DEST_DIR}/voicemail.conf + get_checksum voicemail_conf $file + + local isempty=1 + if [ -z ${voice_contextlist} ] ; then + local isempty=2 + rm -f $file + else + echo "${asteriskuci_gen}[general]" > $file + for i in ${voicegeneral_list} ; do + eval value=\${voice_$i} + if [ ! -z "$value" ] ; then + echo "$i=$value" >> $file + fi + done + echo "${N}[zonemessages]" >> $file + echo "homeloc=${asterisk_zone}| Q IMp" >> $file + echo "${voicezone_list}" >> $file + for i in ${voice_contextlist} ; do + echo "${N}[$i]" >> $file + eval "local cursection=\"\${voicebox_section_${i}}\"" + echo "$cursection" >> $file + eval unset voicebox_section_${i} + done + unset voice_contexts + fi + check_checksum "$voicemail_conf" "$file" || ast_voicemail_restart=$isempty +} + +handle_voicegeneral() { + option_cb() { + if valid_voicemail $1 $2 ; then + eval voice_$1="$2" + else + logerror "Invalid general voice option: $1" + fi + } +} + +handle_voicemail() { + check_add voicebox + voicebox_context=${1%[-_]*} + if [ ${voicebox_context} == $1 ] ; then + voicebox_context=default + fi + voicebox_number=${1#*[-_]} + option_cb() { + case $1 in + zone) voicebox_tz="$2" ;; + *) + if valid_voicebox $1 $2 ; then + eval voicebox_$1="$2" + else + logerror "Invalid voicebox option: $1" + fi + esac + } +} + +# Locality options for voicemail + +check_add_voicezone() { + if [ ! -z "${voicezone_name}" ] ; then + [ -z "${voicezone_zone}" ] && voicezone_zone=${asterisk_zone} + if [ -z "${voicezone_message}" ] ; then + voicezone_message="Q IMp" + else + voicezone_message=`echo "$voicezone_message"|tr \" \'` + fi + append voicezone_list "${voicezone_name}=${voicezone_zone}|${voicezone_message}" "${N}" + fi + unset voicezone_name + unset voicezone_zone + unset voicezone_message +} + +handle_voicezone() { + voicezone_name=$1 + option_cb() { + case $1 in + zone) voicezone_zone="$2" ;; + message) voicezone_message="$2" ;; + *) logerror "Invalid voicezone option: $1" + esac + } +} + +handle_dialplanvoice() { + check_add dialplanvoice + option_cb() { + case $1 in + dialplan|extension|voicecontext|voicebox) + eval "dial_voice_$1=\"$2\"" ;; + *) logerror "Invalid option: $1 for dialplanvoice" + esac + } +} + +check_add_dialplanvoice() { + if [ ! -z "${dial_voice_dialplan}" -a ! -z "${dial_voice_extension}" ] ; then + local ext="exten => ${dial_voice_extension}," + [ -z ${dial_voice_voicebox} ] && dial_voice_voicebox=default + if [ -z ${dial_voice_voicebox} ] ; then + logerror "Expecting voicebox for ${dial_voice_dialplan}/${dial_voice_extension}" + else + check_add_context ${dial_voice_dialplan} + local voiceext="${dial_voice_voicebox}@${dial_voice_voicecontext}" + enable_voicemail + append dialplan_context_${dial_voice_dialplan} "${ext}1,VoiceMailMain(${voiceext})" "${N}" + fi + fi + for i in dialplan extension voicecontext voicebox ; do + eval "unset dial_voice_$i" + done +} + +add_dialplan_voice() { + local context=$1 + logdebug 1 "Adding Dialplan voice $1 $2" + check_add_context "$context" + local ext="exten => $2," + enable_voicemail + append dialplan_context_${context} "${ext}1,VoiceMailMain($3)" "${N}" +} + + + +reload_voicemail() astcmd "module reload app_voicemail.so" +unload_voicemail() astcmd "module unload app_voicemail.so" + + +# vim: ts=2 sw=2 noet foldmethod=indent diff --git a/contrib/asterisk-xip/patches/011-Makefile-main.patch b/contrib/asterisk-xip/patches/011-Makefile-main.patch new file mode 100644 index 0000000000..dbb5c44fee --- /dev/null +++ b/contrib/asterisk-xip/patches/011-Makefile-main.patch @@ -0,0 +1,12 @@ +diff -Nru asterisk-1.4.22.org/main/Makefile asterisk-1.4.22/main/Makefile +--- asterisk-1.4.22.org/main/Makefile 2008-07-18 18:15:41.000000000 +0200 ++++ asterisk-1.4.22/main/Makefile 2008-11-29 14:58:13.000000000 +0100 +@@ -144,7 +144,7 @@ + ifneq ($(findstring chan_h323,$(MENUSELECT_CHANNELS)),) + $(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(ASTLINK) $(AST_EMBED_LDFLAGS) $(ASTLDFLAGS) $^ buildinfo.o $(AST_LIBS) $(AST_EMBED_LIBS) + else +- $(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(ASTLINK) $(AST_EMBED_LDFLAGS) $(ASTLDFLAGS) $(H323LDFLAGS) $^ buildinfo.o $(AST_LIBS) $(AST_EMBED_LIBS) $(H323LDLIBS) ++ $(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(ASTLINK) $(AST_EMBED_LDFLAGS) $(ASTLDFLAGS) $(H323LDFLAGS) $^ buildinfo.o $(AST_LIBS) $(AST_EMBED_LIBS) $(H323LDLIBS) + endif + $(CMD_PREFIX) $(ASTTOPDIR)/build_tools/strip_nonapi $@ || rm $@ + diff --git a/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch b/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch new file mode 100644 index 0000000000..12b64d758c --- /dev/null +++ b/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch @@ -0,0 +1,12 @@ +diff -Nru asterisk-1.4.22.org/channels/chan_iax2.c asterisk-1.4.22/channels/chan_iax2.c +--- asterisk-1.4.22.org/channels/chan_iax2.c 2008-09-02 20:14:57.000000000 +0200 ++++ asterisk-1.4.22/channels/chan_iax2.c 2008-11-29 15:00:00.000000000 +0100 +@@ -1815,7 +1815,7 @@ + last++; + else + last = s; +- snprintf(s2, strlen(s) + 100, "/var/tmp/%s-%ld", last, (unsigned long)ast_random()); ++ snprintf(s2, strlen(s) + 100, "/tmp/%s-%ld", last, (unsigned long)ast_random()); + res = stat(s, &stbuf); + if (res < 0) { + ast_log(LOG_WARNING, "Failed to stat '%s': %s\n", s, strerror(errno)); diff --git a/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch b/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch new file mode 100644 index 0000000000..9f3d9cd925 --- /dev/null +++ b/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch @@ -0,0 +1,12 @@ +diff -Nru asterisk-1.4.22.org/configure.ac asterisk-1.4.22/configure.ac +--- asterisk-1.4.22.org/configure.ac 2008-09-08 18:26:00.000000000 +0200 ++++ asterisk-1.4.22/configure.ac 2008-11-29 15:01:13.000000000 +0100 +@@ -1319,7 +1319,7 @@ + + AST_EXT_LIB_CHECK([SQLITE], [sqlite], [sqlite_exec], [sqlite.h]) + +-AST_EXT_LIB_CHECK([OPENSSL], [ssl], [ssl2_connect], [openssl/ssl.h], [-lcrypto]) ++AST_EXT_LIB_CHECK([OPENSSL], [ssl], [ssl23_connect], [openssl/ssl.h], [-lcrypto]) + if test "$PBX_OPENSSL" = "1"; + then + AST_EXT_LIB_CHECK([OSPTK], [osptk], [OSPPCryptoDecrypt], [osp/osp.h], [-lcrypto -lssl]) diff --git a/contrib/asterisk-xip/patches/015-spandsp-app_fax.patch b/contrib/asterisk-xip/patches/015-spandsp-app_fax.patch new file mode 100644 index 0000000000..ce2c042b2e --- /dev/null +++ b/contrib/asterisk-xip/patches/015-spandsp-app_fax.patch @@ -0,0 +1,875 @@ +diff -Nru asterisk-1.4.22.org/apps/app_rxfax.c asterisk-1.4.22/apps/app_rxfax.c +--- asterisk-1.4.22.org/apps/app_rxfax.c 1970-01-01 01:00:00.000000000 +0100 ++++ asterisk-1.4.22/apps/app_rxfax.c 2008-11-29 15:02:27.000000000 +0100 +@@ -0,0 +1,376 @@ ++/* ++ * Asterisk -- A telephony toolkit for Linux. ++ * ++ * Trivial application to receive a TIFF FAX file ++ * ++ * Copyright (C) 2003, Steve Underwood ++ * ++ * Steve Underwood <steveu@coppice.org> ++ * ++ * This program is free software, distributed under the terms of ++ * the GNU General Public License ++ */ ++ ++/*** MODULEINFO ++ <depend>spandsp</depend> ++***/ ++ ++#include "asterisk.h" ++ ++ASTERISK_FILE_VERSION(__FILE__, "$Revision:$") ++ ++#include <string.h> ++#include <stdlib.h> ++#include <stdio.h> ++#include <inttypes.h> ++#include <pthread.h> ++#include <errno.h> ++#include <tiffio.h> ++ ++#include <spandsp.h> ++ ++#include "asterisk/lock.h" ++#include "asterisk/file.h" ++#include "asterisk/logger.h" ++#include "asterisk/channel.h" ++#include "asterisk/pbx.h" ++#include "asterisk/module.h" ++#include "asterisk/translate.h" ++#include "asterisk/dsp.h" ++#include "asterisk/manager.h" ++ ++static char *app = "RxFAX"; ++ ++static char *synopsis = "Receive a FAX to a file"; ++ ++static char *descrip = ++" RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the\n" ++"given filename. If the file exists it will be overwritten. The file\n" ++"should be in TIFF/F format.\n" ++"The \"caller\" option makes the application behave as a calling machine,\n" ++"rather than the answering machine. The default behaviour is to behave as\n" ++"an answering machine.\n" ++"Uses LOCALSTATIONID to identify itself to the remote end.\n" ++" LOCALHEADERINFO to generate a header line on each page.\n" ++"Sets REMOTESTATIONID to the sender CSID.\n" ++" FAXPAGES to the number of pages received.\n" ++" FAXBITRATE to the transmition rate.\n" ++" FAXRESOLUTION to the resolution.\n" ++"Returns -1 when the user hangs up.\n" ++"Returns 0 otherwise.\n"; ++ ++#define MAX_BLOCK_SIZE 240 ++ ++static void span_message(int level, const char *msg) ++{ ++ int ast_level; ++ ++ if (level == SPAN_LOG_WARNING) ++ ast_level = __LOG_WARNING; ++ else if (level == SPAN_LOG_WARNING) ++ ast_level = __LOG_WARNING; ++ else ++ ast_level = __LOG_DEBUG; ++ ast_log(ast_level, __FILE__, __LINE__, __PRETTY_FUNCTION__, msg); ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static void t30_flush(t30_state_t *s, int which) ++{ ++ /* TODO: */ ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static void phase_e_handler(t30_state_t *s, void *user_data, int result) ++{ ++ struct ast_channel *chan; ++ t30_stats_t t; ++ char local_ident[21]; ++ char far_ident[21]; ++ char buf[11]; ++ ++ chan = (struct ast_channel *) user_data; ++ if (result == T30_ERR_OK) ++ { ++ t30_get_transfer_statistics(s, &t); ++ t30_get_far_ident(s, far_ident); ++ t30_get_local_ident(s, local_ident); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ ast_log(LOG_DEBUG, "Fax successfully received.\n"); ++ ast_log(LOG_DEBUG, "Remote station id: %s\n", far_ident); ++ ast_log(LOG_DEBUG, "Local station id: %s\n", local_ident); ++ ast_log(LOG_DEBUG, "Pages transferred: %i\n", t.pages_transferred); ++ ast_log(LOG_DEBUG, "Image resolution: %i x %i\n", t.x_resolution, t.y_resolution); ++ ast_log(LOG_DEBUG, "Transfer Rate: %i\n", t.bit_rate); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ manager_event(EVENT_FLAG_CALL, ++ "FaxReceived", "Channel: %s\nExten: %s\nCallerID: %s\nRemoteStationID: %s\nLocalStationID: %s\nPagesTransferred: %i\nResolution: %i\nTransferRate: %i\nFileName: %s\n", ++ chan->name, ++ chan->exten, ++ (chan->cid.cid_num) ? chan->cid.cid_num : "", ++ far_ident, ++ local_ident, ++ t.pages_transferred, ++ t.y_resolution, ++ t.bit_rate, ++ s->rx_file); ++ pbx_builtin_setvar_helper(chan, "REMOTESTATIONID", far_ident); ++ snprintf(buf, sizeof(buf), "%i", t.pages_transferred); ++ pbx_builtin_setvar_helper(chan, "FAXPAGES", buf); ++ snprintf(buf, sizeof(buf), "%i", t.y_resolution); ++ pbx_builtin_setvar_helper(chan, "FAXRESOLUTION", buf); ++ snprintf(buf, sizeof(buf), "%i", t.bit_rate); ++ pbx_builtin_setvar_helper(chan, "FAXBITRATE", buf); ++ } ++ else ++ { ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ ast_log(LOG_DEBUG, "Fax receive not successful - result (%d) %s.\n", result, t30_completion_code_to_str(result)); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ } ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static void phase_d_handler(t30_state_t *s, void *user_data, int result) ++{ ++ struct ast_channel *chan; ++ t30_stats_t t; ++ ++ chan = (struct ast_channel *) user_data; ++ if (result) ++ { ++ t30_get_transfer_statistics(s, &t); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ ast_log(LOG_DEBUG, "Pages transferred: %i\n", t.pages_transferred); ++ ast_log(LOG_DEBUG, "Image size: %i x %i\n", t.width, t.length); ++ ast_log(LOG_DEBUG, "Image resolution %i x %i\n", t.x_resolution, t.y_resolution); ++ ast_log(LOG_DEBUG, "Transfer Rate: %i\n", t.bit_rate); ++ ast_log(LOG_DEBUG, "Bad rows %i\n", t.bad_rows); ++ ast_log(LOG_DEBUG, "Longest bad row run %i\n", t.longest_bad_row_run); ++ ast_log(LOG_DEBUG, "Compression type %i\n", t.encoding); ++ ast_log(LOG_DEBUG, "Image size (bytes) %i\n", t.image_size); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ } ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int rxfax_exec(struct ast_channel *chan, void *data) ++{ ++ int res = 0; ++ char template_file[256]; ++ char target_file[256]; ++ char *s; ++ char *t; ++ char *v; ++ const char *x; ++ int option; ++ int len; ++ int i; ++ fax_state_t fax; ++ int calling_party; ++ int verbose; ++ int samples; ++ ++ struct ast_module_user *u; ++ struct ast_frame *inf = NULL; ++ struct ast_frame outf; ++ ++ int original_read_fmt; ++ int original_write_fmt; ++ ++ uint8_t __buf[sizeof(uint16_t)*MAX_BLOCK_SIZE + 2*AST_FRIENDLY_OFFSET]; ++ uint8_t *buf = __buf + AST_FRIENDLY_OFFSET; ++ ++ if (chan == NULL) ++ { ++ ast_log(LOG_WARNING, "Fax receive channel is NULL. Giving up.\n"); ++ return -1; ++ } ++ ++ span_set_message_handler(span_message); ++ ++ /* The next few lines of code parse out the filename and header from the input string */ ++ if (data == NULL) ++ { ++ /* No data implies no filename or anything is present */ ++ ast_log(LOG_WARNING, "Rxfax requires an argument (filename)\n"); ++ return -1; ++ } ++ ++ calling_party = FALSE; ++ verbose = FALSE; ++ target_file[0] = '\0'; ++ ++ for (option = 0, v = s = data; v; option++, s++) ++ { ++ t = s; ++ v = strchr(s, '|'); ++ s = (v) ? v : s + strlen(s); ++ strncpy((char *) buf, t, s - t); ++ buf[s - t] = '\0'; ++ if (option == 0) ++ { ++ /* The first option is always the file name */ ++ len = s - t; ++ if (len > 255) ++ len = 255; ++ strncpy(target_file, t, len); ++ target_file[len] = '\0'; ++ /* Allow the use of %d in the file name for a wild card of sorts, to ++ create a new file with the specified name scheme */ ++ if ((x = strchr(target_file, '%')) && x[1] == 'd') ++ { ++ strcpy(template_file, target_file); ++ i = 0; ++ do ++ { ++ snprintf(target_file, 256, template_file, 1); ++ i++; ++ } ++ while (ast_fileexists(target_file, "", chan->language) != -1); ++ } ++ } ++ else if (strncmp("caller", t, s - t) == 0) ++ { ++ calling_party = TRUE; ++ } ++ else if (strncmp("debug", t, s - t) == 0) ++ { ++ verbose = TRUE; ++ } ++ } ++ ++ /* Done parsing */ ++ ++ u = ast_module_user_add(chan); ++ ++ if (chan->_state != AST_STATE_UP) ++ { ++ /* Shouldn't need this, but checking to see if channel is already answered ++ * Theoretically asterisk should already have answered before running the app */ ++ res = ast_answer(chan); ++ } ++ ++ if (!res) ++ { ++ original_read_fmt = chan->readformat; ++ if (original_read_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_read_format(chan, AST_FORMAT_SLINEAR); ++ if (res < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n"); ++ return -1; ++ } ++ } ++ original_write_fmt = chan->writeformat; ++ if (original_write_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_write_format(chan, AST_FORMAT_SLINEAR); ++ if (res < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n"); ++ res = ast_set_read_format(chan, original_read_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name); ++ return -1; ++ } ++ } ++ fax_init(&fax, calling_party); ++ if (verbose) ++ fax.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW; ++ x = pbx_builtin_getvar_helper(chan, "LOCALSTATIONID"); ++ if (x && x[0]) ++ t30_set_local_ident(&fax.t30_state, x); ++ x = pbx_builtin_getvar_helper(chan, "LOCALHEADERINFO"); ++ if (x && x[0]) ++ t30_set_header_info(&fax.t30_state, x); ++ t30_set_rx_file(&fax.t30_state, target_file, -1); ++ //t30_set_phase_b_handler(&fax.t30_state, phase_b_handler, chan); ++ t30_set_phase_d_handler(&fax.t30_state, phase_d_handler, chan); ++ t30_set_phase_e_handler(&fax.t30_state, phase_e_handler, chan); ++ t30_set_ecm_capability(&fax.t30_state, TRUE); ++ t30_set_supported_compressions(&fax.t30_state, T30_SUPPORT_T4_1D_COMPRESSION | T30_SUPPORT_T4_2D_COMPRESSION | T30_SUPPORT_T6_COMPRESSION); ++ while (ast_waitfor(chan, -1) > -1) ++ { ++ inf = ast_read(chan); ++ if (inf == NULL) ++ { ++ res = -1; ++ break; ++ } ++ if (inf->frametype == AST_FRAME_VOICE) ++ { ++ if (fax_rx(&fax, inf->data, inf->samples)) ++ break; ++ samples = (inf->samples <= MAX_BLOCK_SIZE) ? inf->samples : MAX_BLOCK_SIZE; ++ len = fax_tx(&fax, (int16_t *) &buf[AST_FRIENDLY_OFFSET], samples); ++ if (len) ++ { ++ memset(&outf, 0, sizeof(outf)); ++ outf.frametype = AST_FRAME_VOICE; ++ outf.subclass = AST_FORMAT_SLINEAR; ++ outf.datalen = len*sizeof(int16_t); ++ outf.samples = len; ++ outf.data = &buf[AST_FRIENDLY_OFFSET]; ++ outf.offset = AST_FRIENDLY_OFFSET; ++ outf.src = "RxFAX"; ++ if (ast_write(chan, &outf) < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno)); ++ break; ++ } ++ } ++ } ++ ast_frfree(inf); ++ } ++ if (inf == NULL) ++ { ++ ast_log(LOG_DEBUG, "Got hangup\n"); ++ res = -1; ++ } ++ if (original_read_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_read_format(chan, original_read_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name); ++ } ++ if (original_write_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_write_format(chan, original_write_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", chan->name); ++ } ++ t30_terminate(&fax.t30_state); ++ } ++ else ++ { ++ ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name); ++ } ++ ast_module_user_remove(u); ++ return res; ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int unload_module(void) ++{ ++ int res; ++ ++ ast_module_user_hangup_all(); ++ ++ res = ast_unregister_application(app); ++ ++ ++ return res; ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int load_module(void) ++{ ++ return ast_register_application(app, rxfax_exec, synopsis, descrip); ++} ++/*- End of function --------------------------------------------------------*/ ++ ++AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial FAX Receive Application"); ++ ++/*- End of file ------------------------------------------------------------*/ +diff -Nru asterisk-1.4.22.org/apps/app_txfax.c asterisk-1.4.22/apps/app_txfax.c +--- asterisk-1.4.22.org/apps/app_txfax.c 1970-01-01 01:00:00.000000000 +0100 ++++ asterisk-1.4.22/apps/app_txfax.c 2008-11-29 15:02:27.000000000 +0100 +@@ -0,0 +1,303 @@ ++/* ++ * Asterisk -- A telephony toolkit for Linux. ++ * ++ * Trivial application to send a TIFF file as a FAX ++ * ++ * Copyright (C) 2003, Steve Underwood ++ * ++ * Steve Underwood <steveu@coppice.org> ++ * ++ * This program is free software, distributed under the terms of ++ * the GNU General Public License ++ */ ++ ++/*** MODULEINFO ++ <depend>spandsp</depend> ++***/ ++ ++#include "asterisk.h" ++ ++ASTERISK_FILE_VERSION(__FILE__, "$Revision:$") ++ ++#include <string.h> ++#include <stdlib.h> ++#include <stdio.h> ++#include <inttypes.h> ++#include <pthread.h> ++#include <errno.h> ++#include <tiffio.h> ++ ++#include <spandsp.h> ++ ++#include "asterisk/lock.h" ++#include "asterisk/file.h" ++#include "asterisk/logger.h" ++#include "asterisk/channel.h" ++#include "asterisk/pbx.h" ++#include "asterisk/module.h" ++#include "asterisk/translate.h" ++ ++static char *app = "TxFAX"; ++ ++static char *synopsis = "Send a FAX file"; ++ ++static char *descrip = ++" TxFAX(filename[|caller][|debug]): Send a given TIFF file to the channel as a FAX.\n" ++"The \"caller\" option makes the application behave as a calling machine,\n" ++"rather than the answering machine. The default behaviour is to behave as\n" ++"an answering machine.\n" ++"Uses LOCALSTATIONID to identify itself to the remote end.\n" ++" LOCALHEADERINFO to generate a header line on each page.\n" ++"Sets REMOTESTATIONID to the receiver CSID.\n" ++"Returns -1 when the user hangs up, or if the file does not exist.\n" ++"Returns 0 otherwise.\n"; ++ ++#define MAX_BLOCK_SIZE 240 ++ ++static void span_message(int level, const char *msg) ++{ ++ int ast_level; ++ ++ if (level == SPAN_LOG_WARNING) ++ ast_level = __LOG_WARNING; ++ else if (level == SPAN_LOG_WARNING) ++ ast_level = __LOG_WARNING; ++ else ++ ast_level = __LOG_DEBUG; ++ ast_log(ast_level, __FILE__, __LINE__, __PRETTY_FUNCTION__, msg); ++} ++/*- End of function --------------------------------------------------------*/ ++ ++#if 0 ++static void t30_flush(t30_state_t *s, int which) ++{ ++ /* TODO: */ ++} ++/*- End of function --------------------------------------------------------*/ ++#endif ++ ++static void phase_e_handler(t30_state_t *s, void *user_data, int result) ++{ ++ struct ast_channel *chan; ++ char far_ident[21]; ++ ++ chan = (struct ast_channel *) user_data; ++ if (result == T30_ERR_OK) ++ { ++ t30_get_far_ident(s, far_ident); ++ pbx_builtin_setvar_helper(chan, "REMOTESTATIONID", far_ident); ++ } ++ else ++ { ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ ast_log(LOG_DEBUG, "Fax send not successful - result (%d) %s.\n", result, t30_completion_code_to_str(result)); ++ ast_log(LOG_DEBUG, "==============================================================================\n"); ++ } ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int txfax_exec(struct ast_channel *chan, void *data) ++{ ++ int res = 0; ++ char source_file[256]; ++ char *s; ++ char *t; ++ char *v; ++ const char *x; ++ int option; ++ int len; ++ fax_state_t fax; ++ int calling_party; ++ int verbose; ++ int samples; ++ ++ struct ast_module_user *u; ++ struct ast_frame *inf = NULL; ++ struct ast_frame outf; ++ ++ int original_read_fmt; ++ int original_write_fmt; ++ ++ uint8_t __buf[sizeof(uint16_t)*MAX_BLOCK_SIZE + 2*AST_FRIENDLY_OFFSET]; ++ uint8_t *buf = __buf + AST_FRIENDLY_OFFSET; ++ ++ if (chan == NULL) ++ { ++ ast_log(LOG_WARNING, "Fax transmit channel is NULL. Giving up.\n"); ++ return -1; ++ } ++ ++ span_set_message_handler(span_message); ++ ++ /* The next few lines of code parse out the filename and header from the input string */ ++ if (data == NULL) ++ { ++ /* No data implies no filename or anything is present */ ++ ast_log(LOG_WARNING, "Txfax requires an argument (filename)\n"); ++ return -1; ++ } ++ ++ calling_party = FALSE; ++ verbose = FALSE; ++ source_file[0] = '\0'; ++ ++ for (option = 0, v = s = data; v; option++, s++) ++ { ++ t = s; ++ v = strchr(s, '|'); ++ s = (v) ? v : s + strlen(s); ++ strncpy((char *) buf, t, s - t); ++ buf[s - t] = '\0'; ++ if (option == 0) ++ { ++ /* The first option is always the file name */ ++ len = s - t; ++ if (len > 255) ++ len = 255; ++ strncpy(source_file, t, len); ++ source_file[len] = '\0'; ++ } ++ else if (strncmp("caller", t, s - t) == 0) ++ { ++ calling_party = TRUE; ++ } ++ else if (strncmp("debug", t, s - t) == 0) ++ { ++ verbose = TRUE; ++ } ++ } ++ ++ /* Done parsing */ ++ ++ u = ast_module_user_add(chan); ++ ++ if (chan->_state != AST_STATE_UP) ++ { ++ /* Shouldn't need this, but checking to see if channel is already answered ++ * Theoretically asterisk should already have answered before running the app */ ++ res = ast_answer(chan); ++ } ++ ++ if (!res) ++ { ++ original_read_fmt = chan->readformat; ++ if (original_read_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_read_format(chan, AST_FORMAT_SLINEAR); ++ if (res < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n"); ++ return -1; ++ } ++ } ++ original_write_fmt = chan->writeformat; ++ if (original_write_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_write_format(chan, AST_FORMAT_SLINEAR); ++ if (res < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n"); ++ res = ast_set_read_format(chan, original_read_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name); ++ return -1; ++ } ++ } ++ fax_init(&fax, calling_party); ++ if (verbose) ++ fax.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW; ++ ++ x = pbx_builtin_getvar_helper(chan, "LOCALSTATIONID"); ++ if (x && x[0]) ++ t30_set_local_ident(&fax.t30_state, x); ++ x = pbx_builtin_getvar_helper(chan, "LOCALHEADERINFO"); ++ if (x && x[0]) ++ t30_set_header_info(&fax.t30_state, x); ++ t30_set_tx_file(&fax.t30_state, source_file, -1, -1); ++ //t30_set_phase_b_handler(&fax.t30_state, phase_b_handler, chan); ++ //t30_set_phase_d_handler(&fax.t30_state, phase_d_handler, chan); ++ t30_set_phase_e_handler(&fax.t30_state, phase_e_handler, chan); ++ t30_set_ecm_capability(&fax.t30_state, TRUE); ++ t30_set_supported_compressions(&fax.t30_state, T30_SUPPORT_T4_1D_COMPRESSION | T30_SUPPORT_T4_2D_COMPRESSION | T30_SUPPORT_T6_COMPRESSION); ++ while (ast_waitfor(chan, -1) > -1) ++ { ++ inf = ast_read(chan); ++ if (inf == NULL) ++ { ++ res = -1; ++ break; ++ } ++ if (inf->frametype == AST_FRAME_VOICE) ++ { ++ if (fax_rx(&fax, inf->data, inf->samples)) ++ break; ++ samples = (inf->samples <= MAX_BLOCK_SIZE) ? inf->samples : MAX_BLOCK_SIZE; ++ len = fax_tx(&fax, (int16_t *) &buf[AST_FRIENDLY_OFFSET], samples); ++ if (len) ++ { ++ memset(&outf, 0, sizeof(outf)); ++ outf.frametype = AST_FRAME_VOICE; ++ outf.subclass = AST_FORMAT_SLINEAR; ++ outf.datalen = len*sizeof(int16_t); ++ outf.samples = len; ++ outf.data = &buf[AST_FRIENDLY_OFFSET]; ++ outf.offset = AST_FRIENDLY_OFFSET; ++ if (ast_write(chan, &outf) < 0) ++ { ++ ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno)); ++ break; ++ } ++ } ++ } ++ ast_frfree(inf); ++ } ++ if (inf == NULL) ++ { ++ ast_log(LOG_DEBUG, "Got hangup\n"); ++ res = -1; ++ } ++ if (original_read_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_read_format(chan, original_read_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name); ++ } ++ if (original_write_fmt != AST_FORMAT_SLINEAR) ++ { ++ res = ast_set_write_format(chan, original_write_fmt); ++ if (res) ++ ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", chan->name); ++ } ++ t30_terminate(&fax.t30_state); ++ } ++ else ++ { ++ ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name); ++ } ++ ast_module_user_remove(u); ++ return res; ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int unload_module(void) ++{ ++ int res; ++ ++ ast_module_user_hangup_all(); ++ ++ res = ast_unregister_application(app); ++ ++ ++ return res; ++} ++/*- End of function --------------------------------------------------------*/ ++ ++static int load_module(void) ++{ ++ return ast_register_application(app, txfax_exec, synopsis, descrip); ++} ++/*- End of function --------------------------------------------------------*/ ++ ++AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial FAX Transmit Application"); ++ ++/*- End of file ------------------------------------------------------------*/ +diff -Nru asterisk-1.4.22.org/build_tools/menuselect-deps.in asterisk-1.4.22/build_tools/menuselect-deps.in +--- asterisk-1.4.22.org/build_tools/menuselect-deps.in 2008-08-14 04:02:15.000000000 +0200 ++++ asterisk-1.4.22/build_tools/menuselect-deps.in 2008-11-29 15:02:27.000000000 +0100 +@@ -23,6 +23,7 @@ + POPT=@PBX_POPT@ + PRI=@PBX_PRI@ + RADIUS=@PBX_RADIUS@ ++SPANDSP=@PBX_SPANDSP@ + SPEEX=@PBX_SPEEX@ + SPEEXDSP=@PBX_SPEEXDSP@ + SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@ +diff -Nru asterisk-1.4.22.org/configure.ac asterisk-1.4.22/configure.ac +--- asterisk-1.4.22.org/configure.ac 2008-09-08 18:26:00.000000000 +0200 ++++ asterisk-1.4.22/configure.ac 2008-11-29 15:02:27.000000000 +0100 +@@ -201,6 +201,7 @@ + AST_EXT_LIB_SETUP([PWLIB], [PWlib], [pwlib]) + AST_EXT_LIB_SETUP([OPENH323], [OpenH323], [h323]) + AST_EXT_LIB_SETUP([RADIUS], [Radius Client], [radius]) ++AST_EXT_LIB_SETUP([SPANDSP], [spandsp Library], [spandsp]) + AST_EXT_LIB_SETUP([SPEEX], [Speex], [speex]) + AST_EXT_LIB_SETUP([SPEEXDSP], [Speexdsp], [speexdsp]) + AST_EXT_LIB_SETUP([SQLITE], [SQLite], [sqlite]) +@@ -1302,6 +1303,8 @@ + + AST_EXT_LIB_CHECK([RADIUS], [radiusclient-ng], [rc_read_config], [radiusclient-ng.h]) + ++AST_EXT_LIB_CHECK([SPANDSP], [spandsp], [fax_init], [spandsp.h], [-ltiff -ljpeg -lz]) ++ + AST_EXT_LIB_CHECK([SPEEX], [speex], [speex_encode], [speex/speex.h], [-lm]) + + # See if the main speex library contains the preprocess functions +diff -Nru asterisk-1.4.22.org/include/asterisk/plc.h asterisk-1.4.22/include/asterisk/plc.h +--- asterisk-1.4.22.org/include/asterisk/plc.h 2006-06-14 16:12:56.000000000 +0200 ++++ asterisk-1.4.22/include/asterisk/plc.h 2008-11-29 15:02:27.000000000 +0100 +@@ -1,18 +1,17 @@ +-/*! \file +- * \brief SpanDSP - a series of DSP components for telephony ++/* ++ * SpanDSP - a series of DSP components for telephony + * + * plc.h + * +- * \author Steve Underwood <steveu@coppice.org> ++ * Written by Steve Underwood <steveu@coppice.org> + * + * Copyright (C) 2004 Steve Underwood + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify +- * it under the terms of the GNU General Public License as published by +- * the Free Software Foundation; either version 2 of the License, or +- * (at your option) any later version. ++ * it under the terms of the GNU General Public License version 2, as ++ * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of +@@ -23,37 +22,36 @@ + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * +- * This version may be optionally licenced under the GNU LGPL licence. +- * +- * A license has been granted to Digium (via disclaimer) for the use of +- * this code. ++ * $Id: plc.h,v 1.15 2007/04/08 08:16:18 steveu Exp $ + */ + ++/*! \file */ + +-#if !defined(_PLC_H_) +-#define _PLC_H_ +- +-#ifdef SOLARIS +-#include <sys/int_types.h> +-#else +-#if defined(__OpenBSD__) || defined( __FreeBSD__) +-#include <inttypes.h> +-#else +-#include <stdint.h> +-#endif +-#endif ++#if !defined(_SPANDSP_PLC_H_) ++#define _SPANDSP_PLC_H_ + + /*! \page plc_page Packet loss concealment + \section plc_page_sec_1 What does it do? +-The packet loss concealment module provides a suitable synthetic fill-in signal, +-to minimise the audible effect of lost packets in VoIP applications. It is not +-tied to any particular codec, and could be used with almost any codec which does not ++The packet loss concealment module provides a synthetic fill-in signal, to minimise ++the audible effect of lost packets in VoIP applications. It is not tied to any ++particular codec, and could be used with almost any codec which does not + specify its own procedure for packet loss concealment. + +-Where a codec specific concealment procedure exists, the algorithm is usually built ++Where a codec specific concealment procedure exists, that algorithm is usually built + around knowledge of the characteristics of the particular codec. It will, therefore, + generally give better results for that particular codec than this generic concealer will. + ++The PLC code implements an algorithm similar to the one described in Appendix 1 of G.711. ++However, the G.711 algorithm is optimised for 10ms packets. Few people use such small ++packets. 20ms is a much more common value, and longer packets are also quite common. The ++algorithm has been adjusted with this in mind. Also, the G.711 approach causes an ++algorithmic delay, and requires significant buffer manipulation when there is no packet ++loss. The algorithm used here avoids this. It causes no delay, and achieves comparable ++quality with normal speech. ++ ++Note that both this algorithm, and the one in G.711 are optimised for speech. For most kinds ++of music a much slower decay on bursts of lost packets give better results. ++ + \section plc_page_sec_2 How does it work? + While good packets are being received, the plc_rx() routine keeps a record of the trailing + section of the known speech signal. If a packet is missed, plc_fillin() is called to produce +@@ -83,7 +81,7 @@ + correct steadily fall. Therefore, the volume of the synthesized signal is made to decay + linearly, such that after 50ms of missing audio it is reduced to silence. + +-- When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the ++- When real speech resumes, an extra 1/4 pitch period of synthetic speech is blended with the + start of the real speech. If the erasure is small, this smoothes the transition. If the erasure + is long, and the synthetic signal has faded to zero, the blending softens the start up of the + real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. +@@ -110,6 +108,9 @@ + the pitch assessment. */ + #define PLC_HISTORY_LEN (CORRELATION_SPAN + PLC_PITCH_MIN) + ++/*! ++ The generic packet loss concealer context. ++*/ + typedef struct + { + /*! Consecutive erased samples */ +@@ -127,12 +128,13 @@ + } plc_state_t; + + +-#ifdef __cplusplus +-extern "C" { ++#if defined(__cplusplus) ++extern "C" ++{ + #endif + +-/*! Process a block of received audio samples. +- \brief Process a block of received audio samples. ++/*! Process a block of received audio samples for PLC. ++ \brief Process a block of received audio samples for PLC. + \param s The packet loss concealer context. + \param amp The audio sample buffer. + \param len The number of samples in the buffer. +@@ -147,13 +149,18 @@ + \return The number of samples synthesized. */ + int plc_fillin(plc_state_t *s, int16_t amp[], int len); + +-/*! Process a block of received V.29 modem audio samples. +- \brief Process a block of received V.29 modem audio samples. ++/*! Initialise a packet loss concealer context. ++ \brief Initialise a PLC context. + \param s The packet loss concealer context. +- \return A pointer to the he packet loss concealer context. */ ++ \return A pointer to the the packet loss concealer context. */ + plc_state_t *plc_init(plc_state_t *s); + +-#ifdef __cplusplus ++/*! Free a packet loss concealer context. ++ \param s The packet loss concealer context. ++ \return 0 for OK. */ ++int plc_release(plc_state_t *s); ++ ++#if defined(__cplusplus) + } + #endif + +diff -Nru asterisk-1.4.22.org/makeopts.in asterisk-1.4.22/makeopts.in +--- asterisk-1.4.22.org/makeopts.in 2008-06-12 21:08:20.000000000 +0200 ++++ asterisk-1.4.22/makeopts.in 2008-11-29 15:02:27.000000000 +0100 +@@ -138,6 +138,9 @@ + RADIUS_INCLUDE=@RADIUS_INCLUDE@ + RADIUS_LIB=@RADIUS_LIB@ + ++SPANDSP_INCLUDE=@SPANDSP_INCLUDE@ ++SPANDSP_LIB=@SPANDSP_LIB@ ++ + SPEEX_INCLUDE=@SPEEX_INCLUDE@ + SPEEX_LIB=@SPEEX_LIB@ + diff --git a/contrib/asterisk-xip/patches/016-iksemel-configure_ac.patch b/contrib/asterisk-xip/patches/016-iksemel-configure_ac.patch new file mode 100644 index 0000000000..0b0b7d5e8b --- /dev/null +++ b/contrib/asterisk-xip/patches/016-iksemel-configure_ac.patch @@ -0,0 +1,12 @@ +diff -Nru asterisk-1.4.22.org/configure.ac asterisk-1.4.22/configure.ac +--- asterisk-1.4.22.org/configure.ac 2008-09-08 18:26:00.000000000 +0200 ++++ asterisk-1.4.22/configure.ac 2008-11-29 15:04:09.000000000 +0100 +@@ -514,7 +514,7 @@ + fi + fi + +-AST_EXT_LIB_CHECK([IKSEMEL], [iksemel], [iks_start_sasl], [iksemel.h]) ++AST_EXT_LIB_CHECK([IKSEMEL], [iksemel], [iks_start_sasl], [iksemel.h], [-lgnutls -lgcrypt -lgpg-error]) + + if test "${PBX_IKSEMEL}" = 1; then + AST_EXT_LIB_CHECK([GNUTLS], [gnutls], [gnutls_bye], [gnutls/gnutls.h], [-lz -lgcrypt -lgpg-error]) diff --git a/contrib/asterisk-xip/patches/017-Makefile-no_march.patch b/contrib/asterisk-xip/patches/017-Makefile-no_march.patch new file mode 100644 index 0000000000..98ec1100bc --- /dev/null +++ b/contrib/asterisk-xip/patches/017-Makefile-no_march.patch @@ -0,0 +1,12 @@ +diff -Nru asterisk-1.4.22.org/Makefile asterisk-1.4.22/Makefile +--- asterisk-1.4.22.org/Makefile 2008-09-08 22:15:42.000000000 +0200 ++++ asterisk-1.4.22/Makefile 2008-11-29 15:05:12.000000000 +0100 +@@ -215,7 +215,7 @@ + endif + + ifneq ($(PROC),ultrasparc) +- 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) diff --git a/contrib/asterisk-xip/patches/023-autoconf-chan_h323.patch b/contrib/asterisk-xip/patches/023-autoconf-chan_h323.patch new file mode 100644 index 0000000000..629bb382a7 --- /dev/null +++ b/contrib/asterisk-xip/patches/023-autoconf-chan_h323.patch @@ -0,0 +1,23 @@ +diff -Nru asterisk-1.4.22.org/acinclude.m4 asterisk-1.4.22/acinclude.m4 +--- asterisk-1.4.22.org/acinclude.m4 2008-07-22 22:49:41.000000000 +0200 ++++ asterisk-1.4.22/acinclude.m4 2008-11-29 15:06:28.000000000 +0100 +@@ -588,6 +588,7 @@ + ;; + esac + AC_MSG_RESULT(${OPENH323_BUILD}) ++ OPENH323_SUFFIX="n_s" + + AC_SUBST([OPENH323_SUFFIX]) + AC_SUBST([OPENH323_BUILD]) +diff -Nru asterisk-1.4.22.org/configure.ac asterisk-1.4.22/configure.ac +--- asterisk-1.4.22.org/configure.ac 2008-09-08 18:26:00.000000000 +0200 ++++ asterisk-1.4.22/configure.ac 2008-11-29 15:06:28.000000000 +0100 +@@ -1259,7 +1259,7 @@ + if test "${HAS_PWLIB:-unset}" != "unset"; then + AST_CHECK_OPENH323_PLATFORM() + +- PLATFORM_PWLIB="pt_${PWLIB_PLATFORM}_r" ++ PLATFORM_PWLIB="pt_${PWLIB_PLATFORM}_r_s" + + AST_CHECK_PWLIB_BUILD([PWLib], [PWLIB], + [Define if your system has the PWLib libraries.], diff --git a/contrib/asterisk-xip/patches/026-gsm-mips.patch b/contrib/asterisk-xip/patches/026-gsm-mips.patch new file mode 100644 index 0000000000..e69de29bb2 --- /dev/null +++ b/contrib/asterisk-xip/patches/026-gsm-mips.patch diff --git a/contrib/asterisk-xip/patches/030-acinclude.patch b/contrib/asterisk-xip/patches/030-acinclude.patch new file mode 100644 index 0000000000..7ca79b64ef --- /dev/null +++ b/contrib/asterisk-xip/patches/030-acinclude.patch @@ -0,0 +1,41 @@ +diff -Nru asterisk-1.4.22.org/acinclude.m4 asterisk-1.4.22/acinclude.m4 +--- asterisk-1.4.22.org/acinclude.m4 2008-07-22 22:49:41.000000000 +0200 ++++ asterisk-1.4.22/acinclude.m4 2008-11-29 15:08:07.000000000 +0100 +@@ -664,7 +664,7 @@ + [assume the C compiler uses GNU ld @<:@default=no@:>@])], + [test "$withval" = no || with_gnu_ld=yes], + [with_gnu_ld=no]) +-AC_REQUIRE([AST_PROG_SED])dnl ++AC_REQUIRE([AC_PROG_SED])dnl + AC_REQUIRE([AC_PROG_CC])dnl + AC_REQUIRE([AC_CANONICAL_HOST])dnl + AC_REQUIRE([AC_CANONICAL_BUILD])dnl +@@ -769,28 +769,6 @@ + AC_SUBST([EGREP]) + ])]) # AST_PROG_EGREP + +-# AST_PROG_SED +-# ----------- +-# Check for a fully functional sed program that truncates +-# as few characters as possible. Prefer GNU sed if found. +-AC_DEFUN([AST_PROG_SED], +-[AC_CACHE_CHECK([for a sed that does not truncate output], ac_cv_path_SED, +- [dnl ac_script should not contain more than 99 commands (for HP-UX sed), +- dnl but more than about 7000 bytes, to catch a limit in Solaris 8 /usr/ucb/sed. +- ac_script=s/aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb/ +- for ac_i in 1 2 3 4 5 6 7; do +- ac_script="$ac_script$as_nl$ac_script" +- done +- echo "$ac_script" | sed 99q >conftest.sed +- $as_unset ac_script || ac_script= +- _AC_PATH_PROG_FEATURE_CHECK(SED, [sed gsed], +- [_AC_FEATURE_CHECK_LENGTH([ac_path_SED], [ac_cv_path_SED], +- ["$ac_path_SED" -f conftest.sed])])]) +- SED="$ac_cv_path_SED" +- AC_SUBST([SED])dnl +- rm -f conftest.sed +-])# AST_PROG_SED +- + dnl @synopsis ACX_PTHREAD([ACTION-IF-FOUND[, ACTION-IF-NOT-FOUND]]) + dnl + dnl @summary figure out how to build C programs using POSIX threads diff --git a/contrib/asterisk-xip/patches/035-main-asterisk-uclibc-daemon.patch b/contrib/asterisk-xip/patches/035-main-asterisk-uclibc-daemon.patch new file mode 100644 index 0000000000..262bed4a74 --- /dev/null +++ b/contrib/asterisk-xip/patches/035-main-asterisk-uclibc-daemon.patch @@ -0,0 +1,42 @@ +diff -Nru asterisk-1.4.22.org/main/asterisk.c asterisk-1.4.22/main/asterisk.c +--- asterisk-1.4.22.org/main/asterisk.c 2008-07-26 17:31:21.000000000 +0200 ++++ asterisk-1.4.22/main/asterisk.c 2008-12-20 22:49:58.000000000 +0100 +@@ -2935,7 +2935,38 @@ + #if HAVE_WORKING_FORK + if (ast_opt_always_fork || !ast_opt_no_fork) { + #ifndef HAVE_SBIN_LAUNCHD ++#ifndef __UCLIBC__ + daemon(1, 0); ++#else ++/* ++ workaround for uClibc-0.9.29 mipsel bug: ++ recursive mutexes do not work if uClibc daemon() function has been called, ++ if parent thread locks a mutex ++ the child thread cannot acquire a lock with the same name ++ (same code works if daemon() is not called) ++ but duplication of uClibc daemon.c code in here does work. ++*/ ++ int fd; ++ switch (fork()) { ++ case -1: ++ exit(1); ++ case 0: ++ break; ++ default: ++ _exit(0); ++ } ++ if (setsid() == -1) ++ exit(1); ++ if (fork()) ++ _exit(0); ++ if ((fd = open("/dev/null", O_RDWR, 0)) != -1) { ++ dup2(fd, STDIN_FILENO); ++ dup2(fd, STDOUT_FILENO); ++ dup2(fd, STDERR_FILENO); ++ if (fd > 2) ++ close(fd); ++ } ++#endif + ast_mainpid = getpid(); + /* Blindly re-write pid file since we are forking */ + unlink(ast_config_AST_PID); |