contrib/package: add asterisk14-xip package to prepare Asterisk/LuCI/UCI integration

36 files changed, 6497 insertions, 0 deletions

diff --git a/contrib/asterisk-xip/Makefile b/contrib/asterisk-xip/Makefile
new file mode 100644
index 000000000..3eff7aa96
--- /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 000000000..9d046c42d
--- /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 000000000..64a7d906c
--- /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  $(cat /var/run/asterisk.pid) 2>&- >&-
+}
+
+# vim:ts=2 sw=2
diff --git a/contrib/asterisk-xip/files/macros/clock.conf b/contrib/asterisk-xip/files/macros/clock.conf
new file mode 100644
index 000000000..3250b396a
--- /dev/null
+++ b/contrib/asterisk-xip/files/macros/clock.conf
@@ -0,0 +1,23 @@
+; Talking clock by Michael Geddes
+; Borrowed from bits here and there.
+[macro-talkingclock] ; (TimeFormat, DateFormat, Zone)
+exten => s,1,Answer
+exten => s,n,set(tcTimeFormat=${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} < ${FutureTime} ]?waitforit:)
+exten => s,n,playback(beep)
+exten => s,n,goto(again)
diff --git a/contrib/asterisk-xip/files/macros/lastcall.conf b/contrib/asterisk-xip/files/macros/lastcall.conf
new file mode 100644
index 000000000..5d178617f
--- /dev/null
+++ b/contrib/asterisk-xip/files/macros/lastcall.conf
@@ -0,0 +1,78 @@
+; Last-Called number storage and calling.
+; Author: Michael Geddes aka FrogOnWheels
+
+[macro-lastcallstore] ; (Number , EntryType, BufferSize)
+exten => s,1,set(lcsName=lastcall)
+exten => s,n,set(lcsCount=10)
+exten => s,n,GotoIf($["${ARG2}" = ""]?blankarg)
+exten => s,n,GotoIf($["${ARG2}" = "lastcall"]?blankarg)
+exten => s,n,Set(lcsName=lastcall_${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 000000000..ce12c82dc
--- /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 000000000..d90f9d9cd
--- /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 000000000..8966cb844
--- /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 000000000..1fd8f99b9
--- /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 000000000..70ef7c546
--- /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 000000000..eef930bd3
--- /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 000000000..e336570ef
--- /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 ${feature_park_context}
+		enable_module app_parkandannounce
+		enable_format gsm
+	fi
+}
+
+
+reload_feature() astcmd "module reload res_features.so"
+unload_feature() astcmd "module unload res_features.so"
+
+# vim: ts=2 sw=2 noet foldmethod=indent
diff --git a/contrib/asterisk-xip/files/uci/featureconf.txt b/contrib/asterisk-xip/files/uci/featureconf.txt
new file mode 100644
index 000000000..a47b223da
--- /dev/null
+++ b/contrib/asterisk-xip/files/uci/featureconf.txt
@@ -0,0 +1,25 @@
+See asterisk doco
+featurepark
+	parkext
+	parkpos
+	context
+	parkingtime
+	courtesytone
+	parkedplay
+	adsipark
+	findslot
+	parkedmusicclass
+	transferdigittimeout
+	xfersound
+	xferfailsound
+	pickupexten
+	featuredigittimeout
+	atxfernoanswertimeout
+
+featuremap
+	blindxfer
+	disconnect
+	automon
+	atxfer
+	parkcall
+
diff --git a/contrib/asterisk-xip/files/uci/lastcall b/contrib/asterisk-xip/files/uci/lastcall
new file mode 100755
index 000000000..c5ec6e1c4
--- /dev/null
+++ b/contrib/asterisk-xip/files/uci/lastcall
@@ -0,0 +1,119 @@
+#!/bin/sh
+#
+# Author: Michael Geddes  <michael at frog dot wheelycreek dot net>
+# Copyright 2008 Michael Geddes
+# Licensed under GPL
+
+ast_add_module lastcall
+
+init_lastcall() {
+ #
+ ast_enable_type calllist
+ ast_enable_type calllistdefault
+}
+
+check_add_calllist() {
+	local context=${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}\"" "$N"
+		append_include "macros/lastcall.conf"
+		dialplan_do_add_lastcall=1
+		enable_module app_macro
+		enable_module func_callerid
+		enable_module app_sayunixtime
+		enable_module app_playback
+		enable_module func_db
+		enable_format gsm
+	fi
+}
+# vim: ts=2 sw=2 noet foldmethod=indent
diff --git a/contrib/asterisk-xip/files/uci/lastcall.txt b/contrib/asterisk-xip/files/uci/lastcall.txt
new file mode 100644
index 000000000..cd86be3f1
--- /dev/null
+++ b/contrib/asterisk-xip/files/uci/lastcall.txt
@@ -0,0 +1,15 @@
+
+calllistdefault
+	dialplan  - Override default dialplan for calllist extensions to be added to.
+	length    - Default Length of list
+	dialcontext - Default Context for dialing
+
+calllist {name} - Define a call list context
+	extension - Extension to review list
+	dialplan  - Dialplan for extension to be added to (default to extensions)
+	length    - Length of list
+	listname  - Name of list to use
+	tagname   - Sound file to use for name
+	dialcontext - Context to use when dialing
+	calltype  - Type of context for the wrapper (required when calling it) macro|gosub
+
diff --git a/contrib/asterisk-xip/files/uci/meetmeconf b/contrib/asterisk-xip/files/uci/meetmeconf
new file mode 100755
index 000000000..d70016148
--- /dev/null
+++ b/contrib/asterisk-xip/files/uci/meetmeconf
@@ -0,0 +1,107 @@
+#!/bin/sh
+# Meetme.conf (conference)
+
+ast_add_conf meetme
+init_meetmeconf() {
+	ast_add_reload meetme
+	ast_enable_type meetmegeneral
+	ast_enable_type meetme
+	ast_enable_type dialplanmeetme
+}
+create_meetmeconf() {
+	file=${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 000000000..fc24e389c
--- /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 000000000..d8ea6114a
--- /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 000000000..7b083b6db
--- /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 000000000..9963108cf
--- /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 000000000..232174062
--- /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 000000000..f3f072020
--- /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 000000000..ce2da498c
--- /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 000000000..20e36065f
--- /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 000000000..9d4cdf360
--- /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 000000000..83151f0b9
--- /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 000000000..dbb5c44fe
--- /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 $@
+ 
diff --git a/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch b/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch
new file mode 100644
index 000000000..12b64d758
--- /dev/null
+++ b/contrib/asterisk-xip/patches/013-chan_iax2-tmp_path.patch
@@ -0,0 +1,12 @@
+diff -Nru asterisk-1.4.22.org/channels/chan_iax2.c asterisk-1.4.22/channels/chan_iax2.c
+--- asterisk-1.4.22.org/channels/chan_iax2.c	2008-09-02 20:14:57.000000000 +0200
++++ asterisk-1.4.22/channels/chan_iax2.c	2008-11-29 15:00:00.000000000 +0100
+@@ -1815,7 +1815,7 @@
+ 		last++;
+ 	else
+ 		last = s;
+-	snprintf(s2, strlen(s) + 100, "/var/tmp/%s-%ld", last, (unsigned long)ast_random());
++	snprintf(s2, strlen(s) + 100, "/tmp/%s-%ld", last, (unsigned long)ast_random());
+ 	res = stat(s, &stbuf);
+ 	if (res < 0) {
+ 		ast_log(LOG_WARNING, "Failed to stat '%s': %s\n", s, strerror(errno));
diff --git a/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch b/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch
new file mode 100644
index 000000000..9f3d9cd92
--- /dev/null
+++ b/contrib/asterisk-xip/patches/014-openssl-configure_ac.patch
@@ -0,0 +1,12 @@
+diff -Nru asterisk-1.4.22.org/configure.ac asterisk-1.4.22/configure.ac
+--- asterisk-1.4.22.org/configure.ac	2008-09-08 18:26:00.000000000 +0200
++++ asterisk-1.4.22/configure.ac	2008-11-29 15:01:13.000000000 +0100
+@@ -1319,7 +1319,7 @@
+ 
+ AST_EXT_LIB_CHECK([SQLITE], [sqlite], [sqlite_exec], [sqlite.h])
+ 
+-AST_EXT_LIB_CHECK([OPENSSL], [ssl], [ssl2_connect], [openssl/ssl.h], [-lcrypto])
++AST_EXT_LIB_CHECK([OPENSSL], [ssl], [ssl23_connect], [openssl/ssl.h], [-lcrypto])
+ if test "$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 000000000..ce2c042b2
--- /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 000000000..0b0b7d5e8
--- /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 "${PBX_IKSEMEL}" = 1; then
+    AST_EXT_LIB_CHECK([GNUTLS], [gnutls], [gnutls_bye], [gnutls/gnutls.h], [-lz -lgcrypt -lgpg-error])
diff --git a/contrib/asterisk-xip/patches/017-Makefile-no_march.patch b/contrib/asterisk-xip/patches/017-Makefile-no_march.patch
new file mode 100644
index 000000000..98ec1100b
--- /dev/null
+++ b/contrib/asterisk-xip/patches/017-Makefile-no_march.patch
@@ -0,0 +1,12 @@
+diff -Nru asterisk-1.4.22.org/Makefile asterisk-1.4.22/Makefile
+--- asterisk-1.4.22.org/Makefile	2008-09-08 22:15:42.000000000 +0200
++++ asterisk-1.4.22/Makefile	2008-11-29 15:05:12.000000000 +0100
+@@ -215,7 +215,7 @@
+ endif
+ 
+ ifneq ($(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 000000000..629bb382a
--- /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 000000000..e69de29bb
--- /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 000000000..7ca79b64e
--- /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 000000000..262bed4a7
--- /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);