Version do webrtc no version_1.8.2

version_1.8.2/README.md

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+
+
+# Configuração do PABX versão 1.8 no docker
+
+Inicie a aplicação PABX de maneira fácil e sincronizada.
+
+## Preparando o ambiente no Windows
+
+### Configurando o WSL
+
+- Certifique-se de que o WSL2 está instalado no Windows. Caso não esteja, siga a documentação para realizar a instalação: [Instalação do WSL2](https://learn.microsoft.com/pt-br/windows/wsl/install).
+
+Após a instalação do WSL2, instale uma distribuição Linux.
+
+* Recomendamos a utilização do Debian. Para isso, execute o seguinte comando no PowerShell:
+```bash
+wsl --install -d Debian
+```
+
+* Em seguida, digite o seguinte comando no PowerShell para entrar em uma instância do Debian:
+```bash
+wsl -d Debian
+```
+### Desabilite o Firewall e o Defender do Windows
+Para conseguir realizar o download de alguns repositórios.
+
+
+
+## Clonando o repositório
+
+* Clone o projeto: docker-pabx-app.
+```bash
+git clone http://192.168.115.233:3000/SimplesIP/docker-pabx-app.git
+```
+
+## Deploy
+
+* Entre no diretório que contém os arquivos de inicialização do container.
+```bash
+cd docker-pabx-app/debian
+```
+
+Crie uma cópia do arquivo 'env-example' com o nome de '.env'.
+```bash
+cp env-example .env
+```
+
+Defina as variáveis dentro do arquivo .env:
+- pasta_do_projeto - a pasta onde se encontra o projeto PABX.
+- pasta_do_postgresql11 - onde deverá sincronizar o banco de dados.
+- postgresql_version - por agora, não deve mudar o valor.
+
+### Execute o container com o docker-compose:
+```bash
+sudo docker-compose up -d
+```
+Esse comando tentará compilar e levantar container.
+
+## FAQ
+
+* ### Configurar rota do endereço SIP?
+    Acesse a página `<ip>:8080/config-docker.php`. Essa página vai configurar o IP no arquivo `/etc/sip_generera.conf` e carregá-lo no Asterisk. Não deve usar "localhost", pois o endereço será do container. É necessário usar o endereço do host para acessar esta página config-docker.php.
+
+* ### Para que serve o arquivo `.env` ?
+
+Serve para definir variáveis a serem usadas no compose. Leia o arquivo '.env-example' para mais informações.
+
+* ### Quais arquivos estão sicronizados ?
+
+    Grande parte dos arquivos do projeto estão sincronizados, porém existe os arquivos que não estão sincronizados. Os arquivos não sincronizados serão apagados quando seu container for encerrado e reescritos pelos arquivos da imagem quando o container for iniciado.
+
+    Não sincronizados:
+    - `/etc/asterisk` - Sem sincronização na pasta
+    - `/etc/init.d/cnvrtd` - sem sincronização no arquivo
+    - `/etc/init.d/rtabd` - sem sincronização no arquivo
+    - `/etc/init.d/supervisor` - sem sincronização no arquivo
+
+    Sincronizados:
+     - `/var/lib/postgresql/data` - Sincronizado os arquivo do banco de dados
+     - `/var/www/html/aplicativo` - Sincronizado
+     - `/var/www/html/include` - sincronizado
+     - `/var/lib/asterisk/` - sincronizado
+     - `/hdaux/utilitarios/scripts` - sincronizado
+    
+    Obs:
+    - `/projeto/base` será replicado no container postgres `/base` para poder puxar o banco de dados.
+
+
+### É necessário executar algo depois da build?
+
+    Não, todo o processo é automático. Porém, é preciso acessar `<ip>:8080/config-docker.php`, que fará o reconhecimento do IP e irá defini-lo para o protocolo SIP.
+
+* ### Qual as portas ?
+
+    - `8080` - apache (web)
+    - `5432` - postgresql
+    - `5038` - asterisk (ami)
+    - `5060` - asterisk (sip)
+    - `10000-10030` - asterisk (RTP)
+
+* ### Se o pabx criar arquivos que não estão no projeto gitea?
+    Coloque em `.gitignore` e mande a correção.

version_1.8.2/docker-compose.yml

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+version: '3.7'
+networks:
+    working:
+        driver: bridge
+services:
+    postgres:
+        build: 
+            context: postgres-13
+            args:
+                # definir o id do usuário para ter acesso ao volume
+                - ID_USER_VOLUME=${id_user} 
+        container_name: v1.8.1-webrtc-pg
+        restart: always
+        environment:
+            - POSTGRES_USER=contacte
+            - POSTGRES_PASSWORD=ctepgSQL
+            - POSTGRES_DB=pbx
+        ports:
+            - 5432:5432
+        networks:
+            - working
+        volumes:
+          # copiar a pasta /projeto/base do pabx
+          - ${pasta_do_projeto}/projeto/base:/base
+          # copiar a pasta /var/lib/pgsql/data
+          - ${pasta_do_postgresql}:/var/lib/postgresql/data
+    aplicativo:
+        build:
+            context: pabx
+            args:
+                # usar para compilar dockerfile
+                # DB_HOST DB_PORT DB_BASE - necessário para colocar no crontab
+                # environment compose é para depois do dockerfile
+                - DB_HOST=postgres
+                - DB_PORT=5432
+                - DB_BASE=pbx
+                - ID_USER_VOLUME=${id_user}
+        restart: always
+        container_name: v1.8.1-webrtc
+        working_dir: /var/www/html/
+        ports:
+            - 8080:80
+            - 5038:5038
+            - "5060:5060/tcp"
+            - "5060:5060/udp"
+            - "5061:5061/tcp"
+            - "5061:5061/udp"
+            - "8088:8088/tcp"
+            - "10000-10030:10000-10030/udp"
+            - 4443:443
+            - 8089:8089
+        privileged: true
+        depends_on:
+            - postgres
+        networks:
+            - working
+        links:
+            - postgres
+        volumes:
+            - ./pabx/etc/asterisk/:/etc/asterisk
+            - ./pabx/certificados/:/usr/src/certificates
+            - ${pasta_do_projeto}:/var/www/html/aplicativo
+            - ${pasta_do_projeto}/include:/var/www/html/include
+            - ${pasta_do_projeto}/asterisk/var_lib_asterisk/:/var/lib/asterisk/
+            - ${pasta_do_projeto}/asterisk/hdaux_utilitarios_scripts/:/hdaux/utilitarios/scripts
+

version_1.8.2/env_example

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+
+
+# Pasta aonde está o projeto pabx para ser sincronizado no container
+#
+# Exemplo:
+#pasta_do_projeto=/home/user/source_pabx
+#pasta_do_projeto=c://Usuario/User/source_pabx
+pasta_do_projeto=""
+
+
+# Pasta do host para sincronizar a pasta /var/lib/pgsql/data (versão 13)
+# Isso será usada para manter o banco de dados caso queira levantar
+# outra instância.
+#
+# Essa pasta pode ser enviado ao repositório caso queira compartilhar
+# o banco
+pasta_do_postgresql=""
+
+
+# ID do usuário para ter acesso ao volume
+# Esse id_user é para poder ter acesso ao volume sem a necessidade de se tornar root
+# Será modificado o id dos usuários postgres e pbx 
+# Para saber seu id é o comando 'id -u'.
+id_user="1000"
+
+

version_1.8.2/pabx/Dockerfile

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+FROM debian:bullseye-slim
+
+# Compose manda os valores para conectar ao banco de dados
+# Esses valores são usado para crontab
+# crontab não limpa variáveis de ambiente para chamar script
+# Logo printenv | sed 's/^\(.*\)$/export \1/g' > /env_to_crontab.sh
+# defini as variáveis que vão ser chamado em cada script crontab
+ARG DB_HOST
+ARG DB_PORT
+ARG DB_BASE
+ARG ID_USER_VOLUME
+
+ENV ASTERISK_VERSION 13.38-cert3
+ENV ASTERISK_ADDONS_VERSION 13.18-cert3
+ENV ASTERISK_USER pbx
+ENV ASTERISK_GID pbx
+ENV VERSION_SIMPLESIP v1.8
+ENV APACHE_RUN_USER pbx
+ENV APACHE_RUN_GROUP pbx
+ENV DB_HOST $DB_HOST
+ENV DB_PORT $DB_PORT
+ENV DB_BASE $DB_BASE
+ENV ID_USER_VOLUME $ID_USER_VOLUME
+
+RUN mkdir -p /var/lock/subsys/
+
+# Instalação do asterisk e do PHP 8.2 e asterisk 13.38
+COPY build-asterisk_php.sh /
+RUN chmod 755 /build-asterisk_php.sh
+RUN /build-asterisk_php.sh $ID_USER_VOLUME
+
+COPY etc/asterisk/* /etc/asterisk
+
+# definir as variáveis de ambiente para o cron
+RUN printenv | sed 's/^\(.*\)$/\1/g' > /etc/environment
+COPY crontab.sh /
+RUN bash /crontab.sh
+RUN crontab /etc/crontab
+
+# Script chamado quando acessar a página web ip:8080/config-docker.php
+COPY /config_network /
+RUN chmod 755 /config_network
+RUN chown pbx:pbx /config_network
+
+# apache
+COPY etc/apache2/sites-available/pabx_simples.conf /etc/apache2/sites-available/pabx_simples.conf
+RUN rm -rf /etc/apache2/sites-enabled/*
+RUN ln -s /etc/apache2/sites-available/pabx_simples.conf /etc/apache2/sites-enabled/pabx_simples.conf
+
+
+# colocar os arquivos include_pat do php
+RUN sed -i '745 s/^[;]\? *include_path = .*/include_path = \".:\/var\/www\/html\/include\"/g' /etc/php/8.2/cli/php.ini
+RUN sed -i '745 s/^[;]\? *include_path = .*/include_path = \".:\/var\/www\/html\/include\"/g' /etc/php/8.2/apache2/php.ini
+
+# Executar o apache no usuário pbx
+RUN sed -i 's/APACHE_RUN_USER=.*/APACHE_RUN_USER=pbx/g' /etc/apache2/envvars
+RUN sed -i 's/APACHE_RUN_GROUP=.*/APACHE_RUN_GROUP=pbx/g' /etc/apache2/envvars
+
+COPY docker-entrypoint.sh /
+COPY supervisord.conf /etc/supervisor/supervisord.conf
+
+#RUN sed -i "s/rtpend=.*/rtpend=10030/g" /etc/asterisk/rtp.conf
+#RUN sed -i "s/hostname=.*/hostname=postgres/g" /etc/asterisk/cdr_pgsql.conf
+#RUN sed -i "s/;nocolor = .*/nocolor = yes/g" /etc/asterisk/asterisk.conf
+
+COPY etc/init.d/cnvrtd /etc/init.d/ 
+COPY etc/init.d/rtabd /etc/init.d/
+COPY etc/init.d/supervisor /etc/init.d/
+COPY etc/init.d/functions /etc/init.d/
+
+COPY config-docker.php /var/www/html/
+
+RUN chmod 755 /etc/init.d/rtabd
+RUN chmod 755 /etc/init.d/cnvrtd
+RUN chmod 755 /etc/init.d/supervisor
+
+ENTRYPOINT ["/docker-entrypoint.sh"]
+
+
+CMD ["supervisord","-n","-c","/etc/supervisor/supervisord.conf"]
+
+#docker run -it -d --name v1712 -p 8080:80/tcp -p 5038:5038/tcp -p 5060:5060/udp -p 10000-10050:10000-10050/udp aplicativo
+

version_1.8.2/pabx/build-asterisk_php.sh

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+#!/bin/bash
+PROGNAME=$(basename $0)
+
+if test -z ${ASTERISK_VERSION}; then
+  echo "${PROGNAME}: ASTERISK_VERSION required" >&2
+  exit 1
+fi
+
+set -ex
+
+
+/usr/sbin/groupadd "pbx"
+/usr/sbin/useradd  -u $1 -s "/bin/bash" -d '/var/lib/asterisk/' 'pbx' -g 'pbx'
+
+/usr/sbin/groupadd "aplicacao"
+/usr/sbin/useradd -s "/bin/bash" -d '/tmp' 'aplicacao' -g 'aplicacao'
+
+/usr/sbin/groupadd "simples"
+/usr/sbin/useradd -s "/bin/bash" -m 'simples' -g 'simples'
+
+
+#Remover o source.list e colocar o padrão
+[ -f /etc/apt/sources.list ] && /bin/rm -f /etc/apt/sources.list
+#[ -d /etc/apt/sources.list.d ] && /bin/rm -rf  /etc/apt/souces.list.d/*
+echo "#Principal repositório de Debian" > /etc/apt/sources.list
+echo "deb http://deb.debian.org/debian/ bullseye main contrib non-free" >> /etc/apt/sources.list
+echo "deb-src http://deb.debian.org/debian/ bullseye main contrib non-free" >> /etc/apt/sources.list
+echo ""  >> /etc/apt/sources.list
+echo "deb http://deb.debian.org/debian/ bullseye-updates main contrib non-free" >> /etc/apt/sources.list
+echo "deb-src http://deb.debian.org/debian/ bullseye-updates main contrib non-free" >> /etc/apt/sources.list
+echo ""  >> /etc/apt/sources.list
+echo "# bullseye-updates, to get updates before a point release is made" >> /etc/apt/sources.list
+echo "# see https://www.debian.org/doc/manuals/debian-reference/ch02.en.html#_updates_and_backports" >> /etc/apt/sources.list
+echo "deb http://deb.debian.org/debian-security bullseye-security main" >> /etc/apt/sources.list
+echo "deb-src http://deb.debian.org/debian-security bullseye-security main" >> /etc/apt/sources.list
+
+apt-get -y update
+
+apt-get install lsb-release apt-transport-https ca-certificates software-properties-common -y
+apt-get install wget
+
+wget -O /etc/apt/trusted.gpg.d/php.gpg https://packages.sury.org/php/apt.gpg 
+sh -c 'echo "deb https://packages.sury.org/php/ $(lsb_release -sc) main" > /etc/apt/sources.list.d/php.list'
+
+apt-get -y update
+
+
+apt-get install  -y \
+  supervisor \
+  cron \
+  curl \
+  daemon \
+  libcurl4-openssl-dev \
+  libncurses-dev \
+  libssl-dev \
+  tar \
+  curl \
+  bison \
+  git \
+  vim \
+  apache2 \
+  libjansson-dev \
+  ffmpeg \
+  build-essential \
+  libedit-dev \
+  libsqlite3-dev \
+  uuid-dev \
+  libxml2-dev  \
+  libspeex-dev \
+  libspeexdsp-dev \
+  libogg-dev \
+  libvorbis-dev \
+  libasound2-dev \
+  portaudio19-dev \
+  libcurl4-openssl-dev \
+  xmlstarlet \
+  bison \
+  flex \
+  libpq-dev \
+  unixodbc-dev \
+  libneon27-dev \
+  libgmime-3.0-dev \
+  liblua5.2-dev \
+  liburiparser-dev \
+  libxslt1-dev \
+  libssl-dev  \
+  libmariadb-dev-compat \
+  libmariadb-dev \
+  libbluetooth-dev \
+  libradcli-dev \
+  freetds-dev \
+  libosptk-dev \
+  libjack-jackd2-dev \
+  libcap-dev  \
+  libsnmp-dev \
+  libiksemel-dev \
+  libcorosync-common-dev \
+  libcpg-dev \
+  libcfg-dev \
+  libnewt-dev \
+  libpopt-dev \
+  libical-dev \
+  libspandsp-dev  \
+  libresample1-dev \
+  libc-client2007e-dev \
+  binutils-dev \
+  libsrtp2-dev \
+  libgsm1-dev \
+  graphviz \
+  sudo \
+  zlib1g-dev \
+  libldap2-dev  \
+  bzip2 \
+  libsrtp2-dev \
+  libopus-dev \
+  libvpx-dev \
+  patch \
+  php8.2 \
+  php8.2-cli \
+  php8.2-common \
+  php8.2-dev \
+  php8.2-gd \
+  php8.2-mbstring \
+  php8.2-mysql \
+  php8.2-common \
+  php8.2-snmp \
+  php8.2-soap \
+  php8.2-xml \
+  php8.2-xmlrpc \
+  php8.2-pgsql \
+  php8.2-curl
+
+a2enmod proxy
+a2enmod proxy_wstunnel
+a2enmod ssl
+a2enmod rewrite
+
+### INSTALL CENTRAL SIMPLES IP ###
+mkdir -p /usr/src/asterisk 
+
+cd /usr/src/asterisk
+curl -vsL https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-13.38.3.tar.gz | tar --strip-components 1 -xz
+
+# 1.5 jobs per core works out okay
+: ${JOBS:=$(( $(nproc) + $(nproc) / 2 ))}
+
+mkdir -p /etc/asterisk/ \
+         /var/spool/asterisk/fax
+
+./configure --libdir=/usr/lib64 --with-pjproject-bundled --with-srtp=/usr/include/srtp2
+make menuselect/menuselect menuselect-tree menuselect.makeopts
+
+menuselect/menuselect --disable chan_pjsip menuselect.makeopts
+menuselect/menuselect --enable chan_sip menuselect.makeopts
+menuselect/menuselect --enable cdr_csv menuselect.makeopts
+menuselect/menuselect --enable func_cdr menuselect.makeopts
+menuselect/menuselect --enable cdr_pgsql menuselect.makeopts
+menuselect/menuselect --enable codec_opus menuselect.makeopts
+menuselect/menuselect --enable res_format_attr_opus menuselect.makeopts
+menuselect/menuselect --enable res_crypto menuselect.makeopts
+menuselect/menuselect --enable res_http_websocket menuselect.makeopts
+menuselect/menuselect --enable res_ari menuselect.makeopts
+menuselect/menuselect --enable res_stasis menuselect.makeopts
+menuselect/menuselect --enable res_stun_monitor menuselect.makeopts
+menuselect/menuselect --enable res_phoneprov menuselect.makeopts
+menuselect/menuselect --enable res_odbc menuselect.makeopts
+menuselect/menuselect --enable res_srtp menuselect.makeopts
+
+# we don't need any sounds in docker, they will be mounted as volume
+menuselect/menuselect --disable-category MENUSELECT_CORE_SOUNDS menuselect.makeopts
+menuselect/menuselect --disable-category MENUSELECT_MOH menuselect.makeopts
+menuselect/menuselect --disable-category MENUSELECT_EXTRA_SOUNDS menuselect.makeopts
+
+make -j ${JOBS} all
+make install
+
+# copy default configs
+# cp /usr/src/asterisk/configs/basic-pbx/*.conf /etc/asterisk/
+make dist-clean
+chown -R pbx:pbx  /var/*/asterisk \
+                           /usr/*/asterisk \
+                           /usr/lib64/asterisk
+chmod -R 750 /var/spool/asterisk
+
+cd /
+
+
+#sed -i 's/;nocolor = yes/nocolor = yes/g' /etc/asterisk/asterisk.conf 
+
+echo "/usr/lib64/" > /etc/ld.so.conf.d/local.conf
+ldconfig
+
+apt-get  clean 
+apt-get autoremove -y
+
+
+###    CONFIGURANDO O PHP    ###
+sed -i 's/^error_reporting = .*/error_reporting = E_ALL | E_STRICT/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^error_reporting = .*/error_reporting = E_ALL | E_STRICT/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/display_errors = .*/display_errors = Off/g' /etc/php/8.2/cli/php.ini
+sed -i 's/display_errors = .*/display_errors = Off/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/^display_startup_errors = .*/display_startup_errors = Off/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^display_startup_errors = .*/display_startup_errors = Off/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/^upload_max_filesize = .*/upload_max_filesize = 100M/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^upload_max_filesize = .*/upload_max_filesize = 100M/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/memory_limit = .*/memory_limit = -1/g' /etc/php/8.2/cli/php.ini
+sed -i 's/memory_limit = .*/memory_limit = 256M/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/;error_log = php_errors.*/error_log = \/var\/log\/aplsimples.log/g' /etc/php/8.2/cli/php.ini
+sed -i 's/;error_log = php_errors.*/error_log = \/var\/log\/aplsimples.log/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/^default_charset = .*/default_charset = "iso-8859-1"/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^default_charset = .*/default_charset = "iso-8859-1"/g' /etc/php/8.2/apache2/php.ini
+
+sed -i '745s/^;include_path = .*/include_path = \".:\/var\/www\/html\/include::\/usr\/share\/pear:\/var\/www\/html\/include\/core\"/g' /etc/php/8.2/cli/php.ini
+sed -i '745 s/^;include_path = .*/include_path = \".:\/var\/www\/html\/include::\/usr\/share\/pear:\/var\/www\/html\/include\/core\"/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/^;date.timezone =.*/date.timezone = "America\/Cuiaba"/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^;date.timezone =.*/date.timezone = "America\/Cuiaba"/g' /etc/php/8.2/apache2/php.ini
+
+sed -i 's/^;openssl.cafile=.*/openssl.cafile = \/usr\/src\/certificates\/RootCA.pem/g' /etc/php/8.2/cli/php.ini
+sed -i 's/^;openssl.capath=.*/openssl.capath = \/usr\/src\/certificates/g' /etc/php/8.2/cli/php.ini
+
+sed -i 's/^;openssl.cafile=.*/openssl.cafile = \/usr\/src\/certificates\/RootCA.pem/g' /etc/php/8.2/apache2/php.ini
+sed -i 's/^;openssl.capath=.*/openssl.capath = \/usr\/src\/certificates/g' /etc/php/8.2/apache2/php.ini
+ 
+sed -i 's/^export APACHE_RUN_USER=.*/export APACHE_RUN_USER=pbx/g'  /etc/apache2/envvars
+sed -i 's/^export APACHE_RUN_GROUP=.*/export APACHE_RUN_GROUP=pbx/g' /etc/apache2/envvars
+
+echo "export DB_HOST=${DB_HOST}" >> /etc/apache2/envvars
+
+cd /var/www/html/
+
+exec rm -f /build-asterisk.sh

version_1.8.2/pabx/certificados/RootCA.crt

@@ -0,0 +1,20 @@
+-----BEGIN CERTIFICATE-----
+MIIDMTCCAhmgAwIBAgIUfwDYeruX8TSBfeO3MBY7/p6MuKEwDQYJKoZIhvcNAQEL
+BQAwKDELMAkGA1UEBhMCQlIxGTAXBgNVBAMMEFNpbXBsZXNpcC1XZWJydGMwHhcN
+MjQwNTI0MTU1NjUzWhcNMjcwMzE0MTU1NjUzWjAoMQswCQYDVQQGEwJCUjEZMBcG
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+7+XNNLopBfCxQMTwNYjjz3w+kySrAkNOZdctgnhPvTXWnjuoe8fWQZKtp7Y627U1
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+yaKD/l5z2DOa0wfN9MXKaFY1mMWG2a35SYTZRvo5lsgpOe1oo+2PPlQPXdZsFHBG
+QmN8i9g=
+-----END CERTIFICATE-----

version_1.8.2/pabx/certificados/RootCA.key

@@ -0,0 +1,28 @@
+-----BEGIN PRIVATE KEY-----
+MIIEvwIBADANBgkqhkiG9w0BAQEFAASCBKkwggSlAgEAAoIBAQDuzwVAB5KRM9eB
+7iuVqrT1Yvu2AC7FEG2pVNURieDLUgQvzmQpflbrhq7OJk8YpeFzXHUlI5Qg8RKD
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+uU0W21k3gBFl5F6kfzGepuIdV8f33+uXSZvjzlkI62AVzX13x+b1GjOkF+3ZSZ4C
+jEwnnMmvAgMBAAECggEBAI7FS+E85b5TnRr71ijTdv5hoBTuRQfhy335zqx+nttP
+fFtMbAt2keabjbSHqLAWtBOxCgQbIjsThTtI23a2FLlIg5EfTXfRqkOGPuoPdFPt
+wpyKWSubKocATdtIWCM+l6FIq7rsnOzH4/EccFZ81jIBsgxKvBJE3ZKijEjQkZ0M
+4kDmLXOHxmYaLcwViQREs/JMABkzFAtwZPVu3ipjeICKx1hLf4/BDGeVRwVDb6W+
+AnKhs7CDkm8MBfrhFt1/vvahL6mVkysZ0qY0Moy9h2SE2K0u28Z0iD0OcrOG2CKI
+ez3eVlxTFJ/nzRWKRPN3UR+zfoF2QdSnTrEAhwFp1kECgYEA+oUk+DWwAUwO6AcR
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+WWhMCvvtWMeVOQ0pzio587WXFaHfcbinfSztmzyApKOIiCuTYDfe9sCh2H1DOHG/
+4RbF/I15RK2yVNgaSvlfHrQbhT21CY9SgT7OxVZgmaECgYBD7N5Qan7sHa5erbbG
+BRhoyTUieKHpPZik3EXWwhhJPkc5+hsJLLJvhH2By2FwOn0kCN21SkLgcvruFS4K
+EGa6CuNA5oCEGqoiIVnbz1oqByaYRv3YDxxNDZMH+DNpmvbmKNOvaUBX1HEbMJtG
+PSbAALwrxsh7o6bLnyfK4fcgIQ==
+-----END PRIVATE KEY-----

version_1.8.2/pabx/certificados/RootCA.pem

@@ -0,0 +1,20 @@
+-----BEGIN CERTIFICATE-----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+-----END CERTIFICATE-----

version_1.8.2/pabx/certificados/RootCA.srl

@@ -0,0 +1 @@
+346E4E2447EF354E42823606159D8A94EEC09982

version_1.8.2/pabx/certificados/domains.ext

@@ -0,0 +1,7 @@
+authorityKeyIdentifier=keyid,issuer
+basicConstraints=CA:FALSE
+keyUsage = digitalSignature, nonRepudiation, keyEncipherment, dataEncipherment
+subjectAltName = @alt_names
+[alt_names]
+DNS.1 = localhost
+DNS.2 = 192.168.115.179

version_1.8.2/pabx/certificados/localhost.crt

@@ -0,0 +1,21 @@
+-----BEGIN CERTIFICATE-----
+MIIDjDCCAnSgAwIBAgIUNG5OJEfvNU5CgjYGFZ2KlO7AmYIwDQYJKoZIhvcNAQEL
+BQAwKDELMAkGA1UEBhMCQlIxGTAXBgNVBAMMEFNpbXBsZXNpcC1XZWJydGMwHhcN
+MjQwNTI0MTU1OTI3WhcNMjcwMzE0MTU1OTI3WjB0MQswCQYDVQQGEwJCUjETMBEG
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+e9G+NotQhpE3t07nTp4tSp/sbwp159tkhqJwD4nouic4vwiNXpSsbIkVhVQRdSdw
+sS0nG2hPUw9aM5PWSe29hCuGRMmaspmoXRInLZ1QgyRmI42oFxJ8bc6mNp7uBMF6
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+vI+qehrgAAXM583Krc5c/0SurEXWG5xIV+ysjUKBsw5AKeWg/T8rTvI0Hc1pFARt
+28n7AgMBAAGjYjBgMB8GA1UdIwQYMBaAFG0L3bmPJQd9PjReRfsSkpBWrqbTMAkG
+A1UdEwQCMAAwCwYDVR0PBAQDAgTwMCUGA1UdEQQeMByCCWxvY2FsaG9zdIIPMTky
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+EgjCqoulBfeLhEi0s/N7ewWQGsTf7h5Ay6jhQGWcbJaMW2MV3UhfmrREWGLhWBD9
+gJ9zqU2hTlD0ImXnsz4WEdltzWdP8VNhnWiXOH6lHIQaSHnMw+18JShuCIajBico
+71ZWxnu0g3yTuTGEYnI4Ataqgn7JQzt+xSFXjjOfFdgYLRIn9gRjoA3W1tPYgfU0
+hLF/ZCqY3DxmROvy1pUKS23CkwAcdg+EP5Vmpk1piCn0e/n+kRuGV4uqpqyoODtw
+lN2Fw5aFEgCC79LnSbaGYFCvWY+OLzwtj8BD5UhE/LNDT765mPQFUETPgRYWLSzn
+-----END CERTIFICATE-----

version_1.8.2/pabx/certificados/localhost.csr

@@ -0,0 +1,17 @@
+-----BEGIN CERTIFICATE REQUEST-----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+-----END CERTIFICATE REQUEST-----

version_1.8.2/pabx/certificados/localhost.key

@@ -0,0 +1,28 @@
+-----BEGIN PRIVATE KEY-----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+-----END PRIVATE KEY-----

version_1.8.2/pabx/config-docker.php

@@ -0,0 +1,18 @@
+<?php
+
+# Poderá ser acessado para configurar as informações de rede para 
+# o asterisk. Isso é feito para definir as redes
+# definir tags no asterisk:
+#   externaddr
+#   localnet
+#   media_address
+#   
+# Deve ser acessado com o ip (não 127.0.0.1)
+# <ip>:8080/config-docker.php
+
+$ip = explode(":", $_SERVER['HTTP_HOST'])[0];
+$result_system=system("/config_network $ip", $qwe);
+//echo "$result_system";
+
+
+?>

version_1.8.2/pabx/config_network

@@ -0,0 +1,40 @@
+#!/bin/bash
+
+IP_HOST="$1"
+if [[ -z $IP_HOST ]];then
+	echo "Sem argumento IP. ERROR"
+	exit;
+fi
+
+
+if [[ -n "$(grep "externaddr=" /etc/asterisk/sip_general_customizado.conf )"  ]]
+then
+	sed -i "s/;\?externaddr=.*/externaddr=${IP_HOST}/g" /etc/asterisk/sip_general_customizado.conf
+else
+	echo -e "\nexternaddr=${IP_HOST}" >> /etc/asterisk/sip_general_customizado.conf
+fi
+[[ $? -eq 0 ]] && echo "<h2>[ok] - definido externaddr=${IP_HOST}<br></h2>" || echo "<h2>[false] - externaddr nao definido</h2>"
+
+
+if [[ -n "$(grep "localnet=" /etc/asterisk/sip_general_customizado.conf )" ]]
+then
+	sed -i "s/;\?localnet=.*/localnet=0.0.0.0\/32/g" /etc/asterisk/sip_general_customizado.conf
+else
+	echo "localnet=0.0.0.0/32" >> /etc/asterisk/sip_general_customizado.conf
+fi
+[[ $? -eq 0 ]] &&  echo "<h2>[ok] - definido localnet=0.0.0.0/32<br></h2>" || echo "<h2>[false] - localnet nao definido</h2>"
+
+
+if [[ -n "$(grep "media_address=" /etc/asterisk/sip_general_customizado.conf)" ]]
+then
+	sed -i "s/;\?media_address=.*/media_address=${IP_HOST}/g" /etc/asterisk/sip_general_customizado.conf
+else
+	echo "media_address=${IP_HOST}" >> /etc/asterisk/sip_general_customizado.conf
+fi
+[[ $? -eq 0 ]] &&  echo "<h2>[ok] - definido media_address=${IP_HOST}<br></h2>" || echo "<h2>[false] - media_address nao definido</h2>"
+
+asterisk -rx "sip reload"
+[[ $? -eq 0 ]] && echo "<h2>[ok] - asterisk atualizado<br></h2>" || echo "<h2>[false] - nao carregado no asterisk</h2>"
+
+echo "<h1>Concluido</h1>"
+

version_1.8.2/pabx/crontab.sh

@@ -0,0 +1,40 @@
+#!/bin/bash
+
+echo "#Servicos Auxiliares" >> /etc/crontab
+echo "0  0 * * * root /var/lib/asterisk/scripts/callcenter/reload.sh" >> /etc/crontab
+echo "0 1 * * * root /hdaux/utilitarios/scripts/hd.sh " >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/callcenter/queue_log.php" >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/callcenter/monitora_agentes.php" >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/manutencaoDB/importaBilhetes.php" >> /etc/crontab
+echo "*/30 * * * * root /var/lib/asterisk/scripts/portabilidade/atualiza_cliente.php" >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/workflow/workflowSender.php" >> /etc/crontab
+echo "0 2 * * * root /var/lib/asterisk/scripts/manutencaoDB/manutencaodb.php" >> /etc/crontab
+echo "#*/1 * * * * root /var/lib/asterisk/scripts/integracao/encerraChamada.php" >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/callcenter/atualizar-dacs.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 15 && /var/lib/asterisk/scripts/callcenter/atualizar-dacs.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 30 && /var/lib/asterisk/scripts/callcenter/atualizar-dacs.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 45 && /var/lib/asterisk/scripts/callcenter/atualizar-dacs.php" >> /etc/crontab
+echo "#Transbordo Assistido" >> /etc/crontab
+echo "*/1 * * * * root /var/lib/asterisk/scripts/callcenter/transbordo.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 15 && /var/lib/asterisk/scripts/callcenter/transbordo.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 30 && /var/lib/asterisk/scripts/callcenter/transbordo.php" >> /etc/crontab
+echo "*/1 * * * * root sleep 45 && /var/lib/asterisk/scripts/callcenter/transbordo.php" >> /etc/crontab
+echo "#Backup" >> /etc/crontab
+echo "*/10 * * * * root /var/lib/asterisk/scripts/backup/backup.php" >> /etc/crontab
+echo "#Backup - Shell" >> /etc/crontab
+echo "0 1 * * * root /hdaux/utilitarios/scripts/backup.sh config,bd" >> /etc/crontab
+echo "#VPN" >> /etc/crontab
+echo "*/1 * * * * root /bin/sip_teste_vpn.sh" >> /etc/crontab
+echo "#Remove Caixa Postal" >> /etc/crontab
+echo "0 1 * * * root find /var/spool/asterisk/voicemail/default/ -iname *.WAV -mtime +10 -exec rm -rf {} \\;" >> /etc/crontab
+echo "0 1 * * * root find /var/spool/asterisk/voicemail/default/ -iname *.txt -mtime +10 -exec rm -rf {} \\;" >> /etc/crontab
+echo "#Limpa Maildrop" >> /etc/crontab
+echo "0 6 * * * root for i in \$(ls -1 /var/spool/postfix/maildrop/) ; do /bin/rm -rf /var/spool/postfix/maildrop/\$i ; done" >> /etc/crontab
+echo "#Bloqueio - SIP" >> /etc/crontab
+echo "*/1 * * * * root /hdaux/utilitarios/scripts/bloqueia_autenticacao.sh" >> /etc/crontab
+echo "#NTP" >> /etc/crontab
+echo "*/15 * * * * root /hdaux/utilitarios/scripts/ntp.sh" >> /etc/crontab
+echo "#Logrotate" >> /etc/crontab
+echo "59 23 * * * root /hdaux/utilitarios/scripts/logrotate.sh --force" >> /etc/crontab
+echo "57 23 * * * root /hdaux/utilitarios/scripts/temporizador_de_dados.sh" >> /etc/crontab
+

version_1.8.2/pabx/docker-entrypoint.sh

@@ -0,0 +1,31 @@
+#!/bin/sh
+
+# run as user asterisk by default
+ASTERISK_USER=${ASTERISK_USER:-asterisk}
+
+if [ "$1" = "" ]; then
+  COMMAND="/usr/sbin/asterisk -T -W -U ${ASTERISK_USER} -p -vvvdddf"
+else
+  COMMAND="$@"
+fi
+
+if [ "${ASTERISK_UID}" != "" ] && [ "${ASTERISK_GID}" != "" ]; then
+  # recreate user and group for asterisk
+  # if they've sent as env variables (i.e. to macth with host user to fix permissions for mounted folders
+
+  deluser asterisk && \
+  adduser --gecos "" --no-create-home --uid ${ASTERISK_UID} --disabled-password ${ASTERISK_USER} \
+  || exit
+fi
+
+mkdir -p /var/run/asterisk
+
+chown -R ${ASTERISK_USER}: /var/log/asterisk \
+                           /var/lib/asterisk \
+                           /var/run/asterisk \
+                           /var/spool/asterisk; \
+
+cp /var/www/html/config-docker.php /var/www/html/aplicativo/
+/hdaux/utilitarios/scripts/permissoes_atualizacao.sh
+
+exec ${COMMAND}

version_1.8.2/pabx/etc/apache2/sites-available/pabx_simples.conf

@@ -0,0 +1,100 @@
+<VirtualHost *:80>
+	# The ServerName directive sets the request scheme, hostname and port that
+	# the server uses to identify itself. This is used when creating
+	# redirection URLs. In the context of virtual hosts, the ServerName
+	# specifies what hostname must appear in the request's Host: header to
+	# match this virtual host. For the default virtual host (this file) this
+	# value is not decisive as it is used as a last resort host regardless.
+	# However, you must set it for any further virtual host explicitly.
+	#ServerName www.example.com
+
+	ServerAdmin webmaster@localhost
+	DocumentRoot /var/www/html/aplicativo
+
+	SSLEngine on
+	SSLCertificateFile /usr/src/certificates/localhost.crt
+	SSLCertificateKeyFile /usr/src/certificates/localhost.key
+	#SSLCertificateFile /etc/asterisk/keys/asterisk.crt
+	#SSLCertificateKeyFile /etc/asterisk/keys/asterisk.key
+
+
+	Alias /gravacoes "/var/spool/asterisk/monitor" 
+	<Directory "/var/spool/asterisk/monitor"> 
+		Options Indexes MultiViews 
+		AllowOverride None 
+		Order allow,deny 
+		Require all granted 
+		Allow from 127.0.0.1 
+	</Directory>
+
+	Alias /agenda "/var/www/html/agenda"
+	<Directory "/var/www/html/agenda/"> 
+		Options Indexes MultiViews 
+		AllowOverride None 
+		Order allow,deny 
+		Require all granted 
+		Allow from all 
+	</Directory> 
+
+	Alias /downloads "/hdaux/downloads" 
+	<Directory "/hdaux/downloads/"> 
+		Options Indexes MultiViews
+		AllowOverride None 
+		Order allow,deny 
+		Require all granted 
+		Allow from all 
+	</Directory> 
+
+	Alias /ferramentas "/hdaux/utilitarios/ferramentas"  
+	<Directory "/hdaux/utilitarios/ferramentas/"> 
+		Options Indexes MultiViews 
+		AllowOverride None 
+		Order allow,deny 
+		Require all granted 
+		Allow from all 
+	</Directory>
+
+	Alias /manuais "/hdaux/utilitarios/manuais"  
+	<Directory "/hdaux/utilitarios/manuais/"> 
+		Options Indexes MultiViews 
+		AllowOverride None 
+		Order allow,deny 
+		Require all granted 
+		Allow from all 
+	</Directory> 
+
+
+
+	# Available loglevels: trace8, ..., trace1, debug, info, notice, warn,
+	# error, crit, alert, emerg.
+	# It is also possible to configure the loglevel for particular
+	# modules, e.g.
+	#LogLevel info ssl:warn
+
+	ErrorLog ${APACHE_LOG_DIR}/error.log
+	CustomLog ${APACHE_LOG_DIR}/access.log combined
+
+	# For most configuration files from conf-available/, which are
+	# enabled or disabled at a global level, it is possible to
+	# include a line for only one particular virtual host. For example the
+	# following line enables the CGI configuration for this host only
+	# after it has been globally disabled with "a2disconf".
+	#Include conf-available/serve-cgi-bin.conf
+</VirtualHost>
+
+<VirtualHost *:8089>
+
+ServerAdmin webmaster@localhost
+ServerName localhost
+DocumentRoot /var/www/html
+
+SSLEngine on
+SSLCertificateFile /usr/src/certificates/localhost.crt
+SSLCertificateKeyFile /usr/src/certificates/localhost.key
+#SSLCertificateFile /etc/asterisk/keys/asterisk.crt
+#SSLCertificateKeyFile /etc/asterisk/keys/asterisk.key
+ErrorLog ${APACHE_LOG_DIR}/error.log
+CustomLog ${APACHE_LOG_DIR}/access.log combined
+</VirtualHost>
+
+# vim: syntax=apache ts=4 sw=4 sts=4 sr noet

version_1.8.2/pabx/etc/asterisk/acl.conf

@@ -0,0 +1,80 @@
+;
+; Named Access Control Lists (ACLs)
+;
+; A convenient way to share acl definitions
+;
+; This configuration file is read on startup
+;
+; CLI Commands
+; -----------------------------------------------------------
+;   acl show                         Show all named ACLs configured
+;   acl show <name>                  Show contents of a particular named ACL
+;   reload acl                       Reload configuration file
+;
+; Any configuration that uses ACLs which has been made to be able to use named
+; ACLs will specify a named ACL with the 'acl' option in its configuration in
+; a similar fashion to the usual 'permit' and 'deny' options. Example:
+; acl=my_named_acl
+;
+; Multiple named ACLs can be applied by either comma separating the arguments or
+; just by adding additional ACL lines. Example:
+; acl=my_named_acl
+; acl=my_named_acl2
+;
+; or
+;
+; acl=my_named_acl,my_named_acl2
+;
+; ACLs specified by name are evaluated independently from the ACL specified via
+; permit/deny. In order for an address to pass a given ACL, it must pass both
+; the ACL specified by permit/deny for a given item as well as any named ACLs
+; that were specified.
+;
+;[example_named_acl1]
+;deny=0.0.0.0/0.0.0.0
+;permit=209.16.236.0
+;permit=209.16.236.1
+;
+;[example_named_acl2]
+;permit=0.0.0.0/0.0.0.0
+;deny=10.24.20.171
+;deny=10.24.20.103
+;deny=209.16.236.1
+;
+; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
+; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
+; The following example explains how combining the ACLs works:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+;
+; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
+; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
+; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
+; either, so it passes.
+;
+; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
+; First, example_named_acl1 is evaluated and the address is allowed.
+; However, it is blocked by example_named_acl2, so the address is blocked from the combined
+; ACL.
+;
+; Similarly, the permits/denies in specific configurations that make up an ACL definition
+; are also treated as a separate ACL for evaluation. So if we change the example above to:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+; deny=209.16.236.0
+;
+; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
+; though it passes the two named components.
+;
+;
+; Named ACLs can use ipv6 addresses just like normal ACLs.
+;[ipv6_example_1]
+;deny = ::
+;permit = ::1/128
+;
+;[ipv6_example_2]
+;permit = fe80::21d:bad:fad:2323

version_1.8.2/pabx/etc/asterisk/adsi.conf

@@ -0,0 +1,8 @@
+;
+; Sample ADSI Configuration file
+;
+[intro]
+alignment = center
+greeting => Welcome to the
+greeting => Asterisk
+greeting => Open Source PBX

version_1.8.2/pabx/etc/asterisk/adtranvofr.conf

@@ -0,0 +1,39 @@
+;
+; Voice over Frame Relay (Adtran style)
+;
+; Configuration file
+
+[interfaces]
+;
+; Default language
+;
+;language=en
+;
+; Lines for which we are the user termination.  They accept incoming
+; and outgoing calls.  We use the default context on the first 8 lines
+; used by internal phones.
+;
+context=default
+;user => voice00
+;user => voice01
+;user => voice02
+;user => voice03
+;user => voice04
+;user => voice05
+;user => voice06
+;user => voice07
+; Calls on 16 and 17 come from the outside world, so they get
+; a little bit special treatment
+context=remote
+;user => voice16
+;user => voice17
+;
+; Next we have lines which we only accept calls on, and typically
+; do not send outgoing calls on (i.e. these are where we are the
+; network termination)
+;
+;network => voice08
+;network => voice09
+;network => voice10
+;network => voice11
+;network => voice12

version_1.8.2/pabx/etc/asterisk/agents.conf

@@ -0,0 +1,6 @@
+[general]
+#include agents_general.conf
+#include agents_general_customizado.conf
+#include agents_adicional.conf
+#include agents_usuarios.conf
+#include agents_usuarios_customizado.conf

version_1.8.2/pabx/etc/asterisk/agents_adicional.conf

@@ -0,0 +1,16 @@
+[agents](!)
+;maxlogintries=5
+autologoff=0
+;autologoffunavail=no
+ackcall=no
+;endcall=no
+wrapuptime=0
+musiconhold=agentes
+;goodbye=
+;updatecdr=no
+;group=1
+recordagentcalls=no
+;recordformat=
+;urlprefix=
+;savecallsin=
+custom_beep=beep

version_1.8.2/pabx/etc/asterisk/agents_general.conf

@@ -0,0 +1,2 @@
+;persistentagents=no
+;multiplelogin=no

version_1.8.2/pabx/etc/asterisk/agents_general_customizado.conf

@@ -0,0 +1 @@
+

version_1.8.2/pabx/etc/asterisk/agents_usuarios.conf

@@ -0,0 +1,4 @@
+[1000](agents)
+fullname=admin
+[1001](agents)
+fullname=administrador

version_1.8.2/pabx/etc/asterisk/agents_usuarios_customizado.conf

@@ -0,0 +1 @@
+

version_1.8.2/pabx/etc/asterisk/alarmreceiver.conf

@@ -0,0 +1,91 @@
+;
+; alarmreceiver.conf
+;
+; Sample configuration file for the Asterisk alarm receiver application.
+;
+
+
+[general]
+
+;
+; Specify a timestamp format for the metadata section of the event files
+; Default is %a %b %d, %Y @ %H:%M:%S %Z
+
+timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
+
+;
+; Specify a command to execute when the caller hangs up
+;
+; Default is none
+;
+
+;eventcmd = yourprogram -yourargs ...
+
+;
+; Specify a spool directory for the event files. This setting is required
+; if you want the app to be useful. Event files written to the spool
+; directory will be of the template event-XXXXXX, where XXXXXX is a random
+; and unique alphanumeric string.
+;
+; Default is none, and the events will be dropped on the floor.
+;
+
+eventspooldir = /tmp
+
+;
+; The alarmreceiver app can either log the events one-at-a-time to individual
+; files in the spool directory, or it can store them until the caller
+; disconnects and write them all to one file.
+;
+; The default setting for logindividualevents is no.
+;
+
+logindividualevents = no
+
+;
+; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
+; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
+; receiver by entering digits manually, set this to a reasonable time out
+; like 10000 milliseconds.
+
+fdtimeout = 2000
+
+;
+; The timeout for receiving subsequent DTMF digits is adjustable from
+; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
+; the receiver by entering digits manually, set this to a reasonable time out
+; like 4000 milliseconds.
+;
+
+sdtimeout = 200
+
+;
+; Wait for the connection to settle post-answer. Adjustable from 500 msec. to 10000 msec.
+; The default is 1250 msec.
+;
+
+answait = 1250
+
+; When logging individual events it may be desirable to skip grouping of metadata
+
+;no_group_meta = yes
+
+;
+; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
+; The default is 8192. This shouldn't need to be messed with, but is included
+; just in case there are problems with signal levels.
+;
+
+loudness = 8192
+
+;
+; The db-family setting allows the user to capture statistics on the number of
+; calls, and the errors the alarm receiver sees. The default is for no
+; db-family name to be defined and the database logging to be turned off.
+;
+
+;db-family = yourfamily:
+
+;
+; End of alarmreceiver.conf
+;

version_1.8.2/pabx/etc/asterisk/alsa.conf

@@ -0,0 +1,77 @@
+;
+; Open Sound System Console Driver Configuration File
+;
+[general]
+;
+; Automatically answer incoming calls on the console?  Choose yes if
+; for example you want to use this as an intercom.
+;
+autoanswer=yes
+;
+; Default context (is overridden with @context syntax)
+;
+context=local
+;
+; Default extension to call
+;
+extension=s
+;
+; Default language
+;
+;language=en
+;
+; Default Music on Hold class to use when this channel is placed on hold in
+; the case that the music class is not set on the channel with
+; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
+; putting this one on hold did not suggest a class to use.
+;
+;mohinterpret=default
+;
+; Silence suppression can be enabled when sound is over a certain threshold.
+; The value for the threshold should probably be between 500 and 2000 or so,
+; but your mileage may vary.  Use the echo test to evaluate the best setting.
+;silencesuppression = yes
+;silencethreshold = 1000
+;
+; To set which ALSA device to use, change this parameter
+;input_device=hw:0,0
+;output_device=hw:0,0
+
+;
+; Default mute state (can also be toggled via CLI)
+;mute=true
+
+;
+; If enabled, no audio capture device will be opened.  This is useful on
+; systems where there will be no return audio path, such as overhead pagers.
+;noaudiocapture=true
+
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
+                              ; ALSA channel. Defaults to "no". An enabled jitterbuffer will
+                              ; be used only if the sending side can create and the receiving
+                              ; side can not accept jitter. The ALSA channel can't accept jitter,
+                              ; thus an enabled jitterbuffer on the receive ALSA side will always
+                              ; be used if the sending side can create jitter.
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
+                              ; channel. Two implementations are currently available - "fixed"
+                              ; (with size always equals to jbmax-size) and "adaptive" (with
+                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+; ----------------------------------------------------------------------------------

version_1.8.2/pabx/etc/asterisk/amd.conf

@@ -0,0 +1,31 @@
+;
+; Answering Machine Detection Configuration
+;
+
+[general]
+total_analysis_time = 5000	; Maximum time allowed for the algorithm to decide
+				; on whether the audio represents a HUMAN, or a MACHINE
+silence_threshold = 256		; If the average level of noise in a sample does not reach
+				; this value, from a scale of 0 to 32767, then we will consider
+				; it to be silence.
+
+; Greeting ;
+initial_silence = 2500		; Maximum silence duration before the greeting.
+				; If exceeded, then the result is detection as a MACHINE.
+after_greeting_silence = 800	; Silence after detecting a greeting.
+				; If exceeded, then the result is detection as a HUMAN
+greeting = 1500			; Maximum length of a greeting. If exceeded, then the
+				; result is detection as a MACHINE.
+
+; Word detection ;
+min_word_length = 100		; Minimum duration of Voice to considered as a word
+maximum_word_length = 5000  	; Maximum duration of a single Voice utterance allowed.
+between_words_silence = 50	; Minimum duration of silence after a word to consider
+				; the audio what follows as a new word
+
+maximum_number_of_words = 3	; Maximum number of words in the greeting
+				; If REACHED, then the result is detection as a MACHINE
+				; WARNING: Releases prior to January 1 2016 documented
+				; maximum_number_of_words as 'if exceeded, then MACHINE',
+				; which did not reflect the true functionality.  In Asterisk 14,
+				; this functionality will change to reflect the variables' name.

version_1.8.2/pabx/etc/asterisk/app_mysql.conf

@@ -0,0 +1,24 @@
+; Configuration file for the MYSQL app addon
+
+[general]
+;
+; Nullvalue governs how NULL values are returned from the database.  In
+; previous versions, the special NULL value was returned as the "NULL"
+; string.  We now provide an option for the behavior, configured globally.
+; nullstring  - the string "NULL"
+; emptystring - the string ""
+; null        - unset the variable
+;
+; WARNING: setting nullvalue=null may have undesireable consequences, in
+; particular if you use subroutines in AEL or the LOCAL() variable construct.
+; You have been warned.  Don't complain if you use that setting in combination
+; with Gosub or AEL and get buggy behavior.
+;
+nullvalue = nullstring
+
+; If set, autoclear will destroy allocated statement and connection resources
+; when the channel ends.  For most usage of the MYSQL app, this is what you
+; want, but it's conceivable that somebody is sharing MYSQL connections across
+; multiple channels, in which case, this should be set to 'no'.  Defaults to
+; 'no', as this was the original behavior.
+autoclear=yes

version_1.8.2/pabx/etc/asterisk/app_skel.conf

@@ -0,0 +1,27 @@
+[general]
+games=3
+cheat=no
+
+[sounds]
+prompt=please-enter-your,number,queue-less-than
+wrong_guess=vm-pls-try-again
+right_guess=auth-thankyou
+too_high=high
+too_low=low
+lose=vm-goodbye
+
+[easy]
+max_number=10
+max_guesses=4
+
+[medium]
+max_number=100
+max_guesses=6
+
+[hard]
+max_number=1000
+max_guesses=7
+
+[nightmare]
+max_number=1000
+max_guesses=1

version_1.8.2/pabx/etc/asterisk/ari.conf

@@ -0,0 +1,30 @@
+[general]
+enabled = yes       ; When set to no, ARI support is disabled.
+;pretty = no        ; When set to yes, responses from ARI are
+;                   ; formatted to be human readable.
+;allowed_origins =  ; Comma separated list of allowed origins, for
+;                   ; Cross-Origin Resource Sharing. May be set to * to
+;                   ; allow all origins.
+;auth_realm =       ; Realm to use for authentication. Defaults to Asterisk
+;                   ; REST Interface.
+;
+; Default write timeout to set on websockets. This value may need to be adjusted
+; for connections where Asterisk must write a substantial amount of data and the
+; receiving clients are slow to process the received information. Value is in
+; milliseconds; default is 100 ms.
+;websocket_write_timeout = 100
+
+;[username]
+;type = user        ; Specifies user configuration
+;read_only = no     ; When set to yes, user is only authorized for
+;                   ; read-only requests.
+;
+;password =         ; Crypted or plaintext password (see password_format).
+;
+; password_format may be set to plain (the default) or crypt. When set to crypt,
+; crypt(3) is used to validate the password. A crypted password can be generated
+; using mkpasswd -m sha-512.
+;
+; When set to plain, the password is in plaintext.
+;
+;password_format = plain

version_1.8.2/pabx/etc/asterisk/ast_debug_tools.conf

@@ -0,0 +1,57 @@
+#
+# This file is used by the Asterisk debug tools.
+# Unlike other Asterisk config files, this one is
+# "sourced" by bash and must adhere to bash semantics.
+#
+
+# A list of coredumps and/or coredump search patterns.
+# Bash extended globs are enabled and any resulting files
+# that aren't actually coredumps are silently ignored
+# so you can be liberal with the globs.
+#
+# If your patterns contains spaces be sure to only quote
+# the portion of the pattern that DOESN'T contain wildcard
+# expressions.  If you quote the whole pattern, it won't
+# be expanded and the glob characters will be treated as
+# literals.
+#
+# The exclusion of files ending ".txt" is just for
+# demonstration purposes as non-coredumps will be ignored
+# anyway.
+COREDUMPS=(/tmp/core[-._]asterisk!(*.txt) /tmp/core[-._]$(hostname)!(*.txt))
+
+# Date command for the "running" coredump and tarballs.
+# DATEFORMAT will be executed to get the timestamp.
+# Don't put quotes around the format string or they'll be
+# treated as literal characters.  Also be aware of colons
+# in the output as you can't upload files with colons in
+# the name to Jira.
+#
+# Unix timestamp
+#DATEFORMAT='date +%s.%N'
+#
+# Unix timestamp on *BSD/MacOS after installing coreutils
+#DATEFORMAT='gdate +%s.%N'
+#
+# Readable GMT
+#DATEFORMAT='date -u +%FT%H-%M-%S%z'
+#
+# Readable Local time
+DATEFORMAT='date +%FT%H-%M-%S%z'
+
+# A list of log files and/or log file search patterns using the
+# same syntax as COREDUMPS.
+#
+LOGFILES=(/var/log/asterisk/messages* /var/log/asterisk/queue* \
+	/var/log/asterisk/debug* /var/log/asterisk/security*)
+
+# ast_loggrabber converts POSIX timestamps to readable format
+# using this Python strftime format string.  If not specified
+# or an empty string, no format covnersion is done.
+LOG_DATEFORMAT="%m/%d-%H:%M:%S.%f"
+
+# The timezone to use when converting POSIX timestamps to
+# readable format.  It can be specified in "<continent>/<city>"
+# format or in abbreviation format such as "CST6CDT".  If not
+# specified, the "local" timezone is used.
+# LOG_TIMEZONE=

version_1.8.2/pabx/etc/asterisk/asterisk.adsi

@@ -0,0 +1,158 @@
+;
+; Asterisk default ADSI script
+;
+;
+; Begin with the preamble requirements
+;
+DESCRIPTION "Asterisk PBX"		; Name of vendor
+VERSION 0x00						; Version of stuff
+;SECURITY "_AST"					; Security code
+SECURITY 0X9BDBF7AC				; Security code
+FDN 0x0000000F						; Descriptor number
+
+;
+; Flags
+;
+FLAG "nocallwaiting"
+
+;
+; Predefined strings
+;
+DISPLAY "titles" IS "** Asterisk PBX **"
+DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
+DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
+DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
+DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
+DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
+DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
+DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
+DISPLAY "busy" IS "Busy." JUSTIFY LEFT
+DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
+DISPLAY "cwdisabled" IS "Callwait disabled"
+DISPLAY "empty" IS "asdf"
+
+;
+; Begin soft key definitions
+;
+KEY "callfwd" IS "CallFwd" OR "Call Forward"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "*60"
+	GOTO "offHook"
+ENDKEY
+
+KEY "vmail_OH" IS "VMail" OR "Voicemail"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "vmail" IS "VMail" OR "Voicemail"
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "backspace" IS "BackSpc" OR "Backspace"
+	BACKSPACE
+ENDKEY
+
+KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
+	SENDDTMF "*70"
+	SETFLAG "nocallwaiting"
+	SHOWDISPLAY "cwdisabled" AT 4
+	TIMERCLEAR
+	TIMERSTART 1
+ENDKEY
+
+KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
+	SENDDTMF "*67"
+	SETFLAG "nocallwaiting"
+ENDKEY
+
+;
+; Begin main subroutine
+;
+
+SUB "main" IS
+	IFEVENT NEARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "talkingto" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3
+		SHOWDISPLAY "callnum" AT 4
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT OFFHOOK THEN
+		CLEAR
+		CLEARFLAG "nocallwaiting"
+		CLEARDISPLAY
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail"
+		SHOWKEYS "cidblock"
+		SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
+		GOTO "offHook"
+	ENDIF
+	IFEVENT IDLE THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT CALLERID THEN
+		CLEAR
+;		SHOWDISPLAY "titles" AT 1 NOUPDATE
+;		SHOWDISPLAY "incoming" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3 NOUPDATE
+		SHOWDISPLAY "callnum" AT 4
+	ENDIF
+	IFEVENT RING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "incoming" AT 2
+	ENDIF
+	IFEVENT ENDOFRING THEN
+		SHOWDISPLAY "missedcall" AT 2
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT TIMER THEN
+		CLEAR
+		SHOWDISPLAY "empty" AT 4
+	ENDIF
+ENDSUB
+
+SUB "offHook" IS
+	IFEVENT FARRING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "ringing" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT FARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "talkingto" AT 2
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT BUSY THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "busy" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT REORDER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "reorder" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+ENDSUB
+
+SUB "stableCall" IS
+	IFEVENT REORDER THEN
+		SHOWDISPLAY "callended" AT 2
+	ENDIF
+ENDSUB

version_1.8.2/pabx/etc/asterisk/asterisk.conf

@@ -0,0 +1,114 @@
+[directories](!)
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+astsbindir => /usr/sbin
+
+[options]
+verbose = 45
+;debug = 3
+;alwaysfork = yes		; Same as -F at startup.
+;nofork = yes			; Same as -f at startup.
+;quiet = yes			; Same as -q at startup.
+;timestamp = yes		; Same as -T at startup.
+;execincludes = yes		; Support #exec in config files.
+;console = yes			; Run as console (same as -c at startup).
+;highpriority = yes		; Run realtime priority (same as -p at
+				; startup).
+;initcrypto = yes		; Initialize crypto keys (same as -i at
+				; startup).
+nocolor = yes			; Disable console colors.
+;dontwarn = yes			; Disable some warnings.
+;dumpcore = yes			; Dump core on crash (same as -g at startup).
+;languageprefix = yes		; Use the new sound prefix path syntax.
+;systemname = my_system_name	; Prefix uniqueid with a system name for
+				; Global uniqueness issues.
+;autosystemname = yes		; Automatically set systemname to hostname,
+				; uses 'localhost' on failure, or systemname if
+				; set.
+;mindtmfduration = 80		; Set minimum DTMF duration in ms (default 80 ms)
+				; If we get shorter DTMF messages, these will be
+				; changed to the minimum duration
+;maxcalls = 10			; Maximum amount of calls allowed.
+;maxload = 0.9			; Asterisk stops accepting new calls if the
+				; load average exceed this limit.
+;maxfiles = 1000		; Maximum amount of openfiles.
+;minmemfree = 1			; In MBs, Asterisk stops accepting new calls if
+				; the amount of free memory falls below this
+				; watermark.
+;cache_record_files = yes	; Cache recorded sound files to another
+				; directory during recording.
+;record_cache_dir = /tmp	; Specify cache directory (used in conjunction
+				; with cache_record_files).
+;transmit_silence = yes		; Transmit silence while a channel is in a
+				; waiting state, a recording only state, or
+				; when DTMF is being generated.  Note that the
+				; silence internally is generated in raw signed
+				; linear format. This means that it must be
+				; transcoded into the native format of the
+				; channel before it can be sent to the device.
+				; It is for this reason that this is optional,
+				; as it may result in requiring a temporary
+				; codec translation path for a channel that may
+				; not otherwise require one.
+;transcode_via_sln = yes	; Build transcode paths via SLINEAR, instead of
+				; directly.
+runuser = pbx		; The user to run as.
+rungroup = pbx		; The group to run as.
+;lightbackground = yes		; If your terminal is set for a light-colored
+				; background.
+;forceblackbackground = yes     ; Force the background of the terminal to be
+                                ; black, in order for terminal colors to show
+                                ; up properly.
+;defaultlanguage = en           ; Default language
+documentation_language = en_US	; Set the language you want documentation
+				; displayed in. Value is in the same format as
+				; locale names.
+;hideconnect = yes		; Hide messages displayed when a remote console
+				; connects and disconnects.
+;lockconfdir = no		; Protect the directory containing the
+				; configuration files (/etc/asterisk) with a
+				; lock.
+;stdexten = gosub		; How to invoke the extensions.conf stdexten.
+				; macro - Invoke the stdexten using a macro as
+				;         done by legacy Asterisk versions.
+				; gosub - Invoke the stdexten using a gosub as
+				;         documented in extensions.conf.sample.
+				; Default gosub.
+;live_dangerously = no		; Enable the execution of 'dangerous' dialplan
+				; functions from external sources (AMI,
+				; etc.) These functions (such as SHELL) are
+				; considered dangerous because they can allow
+				; privilege escalation.
+				; Default no
+;entityid=00:11:22:33:44:55	; Entity ID.
+				; This is in the form of a MAC address.
+				; It should be universally unique.
+				; It must be unique between servers communicating
+				; with a protocol that uses this value.
+				; This is currently is used by DUNDi and
+				; Exchanging Device and Mailbox State
+				; using protocols: XMPP, Corosync and PJSIP.
+;rtp_pt_dynamic = 96		; Normally the Dynamic RTP Payload Type numbers
+				; are 96-127, which allow 32 formats. When you
+				; use more and receive the message "No Dynamic
+				; RTP mapping available", extend the dynamic
+				; range by going for 35 (or 0) instead of 96.
+				; This allows 29 (or 64) more formats. 96 is the
+				; default because any number below might be
+				; rejected by a remote implementation; although
+				; no such broken implementation is known, yet.
+
+; Changing the following lines may compromise your security.
+;[files]
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = apache
+;astctl = asterisk.ctl

version_1.8.2/pabx/etc/asterisk/calendar.conf

@@ -0,0 +1,109 @@
+;[calendar1]
+;type = ical              ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/home/jdoe/Calendar/   ; URL to shared calendar (Zimbra example)
+;user = jdoe              ; web username
+;secret = supersecret     ; web password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+; The following fields are available from the ${CALENDAR_EVENT(<field>)} dialplan function:
+;
+; summary     : The VEVENT Summary property or Exchange subject
+; description : The text description of the vent
+; organizer   : The organizer of the event
+; location    : The location field of the event
+; calendar    : The name of the calendar tied to the event
+; uid         : The unique ID for this event
+; start       : Start time of the event
+; end         : The end time of the event
+; busystate   : 0=FREE, 1=TENTATIVE, 2=BUSY
+;
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/60001      ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 123          ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds
+;
+; Channel variables can be set on the notification channel. The format is
+; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan
+; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value
+; of earlier variables in the definition of later ones.
+;
+;setvar = CALLERID(name)=${CALENDAR_EVENT(summary)}
+
+;[calendar2]
+; Note: Support for Exchange Server 2003
+;
+;type = exchange          ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/exchange/jdoe   ; URL to MS Exchange OWA for user (usually includes exchange/user)
+;user = jdoe              ; Exchange username
+;secret = mysecret        ; Exchange password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;[calendar3]
+; Note: Support for Exchange Server 2007+
+;
+;type = ews               ; type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/ews/Exchange.asmx ; URL to MS Exchange EWS
+;user = jdoe              ; Exchange username
+;secret = mysecret        ; Exchange password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds
+
+;[calendar4]
+;type = caldav            ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://www.google.com/calendar/dav/username@gmail.com/events/  ; Main GMail calendar (the trailing slash is significant!)
+;user = jdoe@gmail.com    ; username
+;secret = mysecret        ; password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds

version_1.8.2/pabx/etc/asterisk/ccss.conf

@@ -0,0 +1,205 @@
+;
+; --- Call Completion Supplementary Services ---
+;
+; For more information about CCSS, see the CCSS user documentation
+; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
+;
+
+[general]
+; The cc_max_requests option is a global limit on the number of
+; CC requests that may be in the Asterisk system at any time.
+;
+;cc_max_requests = 20
+;
+; The cc_STATE_devstate variables listed below can be used to change the
+; default mapping of the internal state machine tracking the state of
+; call completion to an Asterisk Device State value. The acceptable values
+; that can be provided are as follows, with a description of what the
+; equivalent device BLF that this maps to:
+;
+;	UNKNOWN      ; Device is valid but channel didn't know state
+;	NOT_INUSE    ; Device is not used
+;	INUSE        ; Device is in use
+;	BUSY         ; Device is busy
+;	INVALID      ; Device is invalid
+;	UNAVAILABLE  ; Device is unavailable
+;	RINGING      ; Device is ringing
+;	RINGINUSE    ; Device is ringing *and* in use
+;	ONHOLD       ; Device is on hold
+;
+; These states are used to generate DEVICE_STATE information that can be
+; included with Asterisk hints for phones to subscribe to the state information
+; or dialplan to check the state using the EXTENSION_STATE() function or
+; the DEVICE_STATE() function.
+;
+; The states are in the format of: "ccss:TECH/ID" so an example of device
+; SIP/3000 making a CallCompletionRequest() could be checked  by looking at
+; DEVICE_STATE(ccss:SIP/3000) or an Asterisk Hint could be generated such as
+;
+; [hint-context]
+; exten => *843000,hint,ccss:SIP/3000
+;
+; and then accessed with EXTENSION_STATE(*843000@hint-context)
+; or subscribed to with a BLF button on a phone.
+;
+; The available state mapping and default values are:
+;
+; cc_available_devstate = NOT_INUSE
+; cc_offered_devstate = NOT_INUSE
+; cc_caller_requested_devstate = NOT_INUSE
+; cc_active_devstate = INUSE
+; cc_callee_ready_devstate = INUSE
+; cc_caller_busy_devstate = ONHOLD
+; cc_recalling_devstate = RINGING
+; cc_complete_devstate = NOT_INUSE
+; cc_failed_devstate = NOT_INUSE
+
+;
+;============================================
+;           PLEASE READ THIS!!!
+; The options described below should NOT be
+; set in this file. Rather, they should be
+; set per-device in a channel driver
+; configuration file.
+;           PLEASE READ THIS!!!
+;===========================================
+;
+; --------------------------------------------------------------------
+;                                Timers
+; --------------------------------------------------------------------
+;There are three configurable timers for all types of CC: the
+;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
+;In addition, when using a generic agent, there is a fourth timer,
+;the cc_recall_timer. All timers are configured in seconds, and the
+;values shown below are the defaults.
+;
+;When a caller is offered CCBS or CCNR, the cc_offer_timer will
+;be started. If the caller does not request CC before the
+;cc_offer_timer expires, then the caller will be unable to request
+;CC for this call.
+;
+;cc_offer_timer = 20
+;
+;Once a caller has requested CC, then either the ccbs_available_timer
+;or the ccnr_available_timer will run, depending on the service
+;requested. The reason why there are two separate timers for CCBS
+;and CCNR is that it is reasonable to want to have a shorter timeout
+;configured for CCBS than for CCNR. If the available timer expires
+;before the called party becomes available, then the CC attempt
+;will have failed and monitoring of the called party will stop.
+;
+;ccbs_available_timer = 4800
+;ccnr_available_timer = 7200
+;
+; When using a generic agent, the original caller is called back
+; when one of the original called parties becomes available. The
+; cc_recall_timer tells Asterisk how long it should let the original
+; caller's phone ring before giving up. Please note that this parameter
+; only affects operation when using a generic agent.
+;
+;cc_recall_timer = 20
+; --------------------------------------------------------------------
+;                                Policies
+; --------------------------------------------------------------------
+; Policy settings tell Asterisk how to behave and what sort of
+; resources to allocate in order to facilitate CC. There are two
+; settings to control the actions Asterisk will take.
+;
+; The cc_agent_policy describes the behavior that Asterisk will
+; take when communicating with the caller during CC. There are
+; three possible options.
+;
+;never:   Never offer CC to the caller. Setting the cc_agent_policy
+;         to this value is the way to disable CC for a call.
+;
+;generic: A generic CC agent is one which uses no protocol-specific
+;         mechanisms to offer CC to the caller. Instead, the caller
+;         requests CC using a dialplan function. Due to internal
+;         restrictions, you should only use a generic CC agent on
+;         phones (i.e. not "trunks"). If you are using phones which
+;         do not support a protocol-specific method of using CC, then
+;         generic CC agents are what you should use.
+;
+;native:  A native CC agent is one which uses protocol-specific
+;         signaling to offer CC to the caller and accept CC requests
+;         from the caller. The supported protocols for native CC
+;         agents are SIP, ISDN ETSI PTP, ISDN ETSI PTMP, and Q.SIG
+;cc_agent_policy=never
+;
+; The cc_monitor_policy describes the behavior that Asterisk will
+; take when communicating with the called party during CC. There
+; are four possible options.
+;
+;never:   Analogous to the cc_agent_policy setting. We will never
+;         attempt to request CC services on this interface.
+;
+;generic: Analogous to the cc_agent_policy setting. We will monitor
+;         the called party's progress using protocol-agnostic
+;         capabilities. Like with generic CC agents, generic CC
+;         monitors should only be used for phones.
+;
+;native:  Analogous to the cc_agent_policy setting. We will use
+;         protocol-specific methods to request CC from this interface
+;         and to monitor the interface for availability.
+;
+;always:  If an interface is set to "always," then we will accept
+;         protocol-specific CC offers from the caller and use
+;         a native CC monitor for the remainder of the CC transaction.
+;         However, if the interface does not offer protocol-specific
+;         CC, then we will fall back to using a generic CC monitor
+;         instead. This is a good setting to use for phones for which
+;         you do not know if they support protocol-specific CC
+;         methodologies.
+;cc_monitor_policy=never
+;
+;
+; --------------------------------------------------------------------
+;                              Limits
+; --------------------------------------------------------------------
+;
+; The use of CC requires Asterisk to potentially use more memory than
+; some administrators would like. As such, it is a good idea to limit
+; the number of CC requests that can be in the system at a given time.
+; The values shown below are the defaults.
+;
+; The cc_max_agents setting limits the number of outstanding CC
+; requests a caller may have at any given time. Please note that due
+; to implementation restrictions, this setting is ignored when using
+; generic CC agents. Generic CC agents may only have one outstanding
+; CC request.
+;
+;cc_max_agents = 5
+;
+; The cc_max_monitors setting limits the number of outstanding CC
+; requests can be made to a specific interface at a given time.
+;
+;cc_max_monitors = 5
+;
+; --------------------------------------------------------------------
+;                            Other
+; --------------------------------------------------------------------
+;
+; When using a generic CC agent, the caller who requested CC will be
+; called back when a called party becomes available. When the caller
+; answers his phone, the administrator may opt to have a macro run.
+; What this macro does is up to the administrator. By default there
+; is no callback macro configured.
+;
+;cc_callback_macro=
+;
+; Alternatively, the administrator may run a subroutine. By default
+; there is no callback subroutine configured.  The subroutine should
+; be specified in the format: [[context,]exten,]priority
+;
+;cc_callback_sub=
+;
+; When using an ISDN phone and a generic CC agent, Asterisk is unable
+; to determine the dialstring that should be used when calling back
+; the original caller. Furthermore, if you desire to use any dialstring-
+; specific options, such as distinctive ring, you must set this
+; configuration option. For non-ISDN phones, it is not necessary to
+; set this, since Asterisk can determine the dialstring to use since
+; it is identical to the name of the calling device. By default, there
+; is no cc_agent_dialstring set.
+;
+;cc_agent_dialstring=

version_1.8.2/pabx/etc/asterisk/cdr.conf

@@ -0,0 +1,167 @@
+;
+; Asterisk Call Detail Record engine configuration
+;
+; CDR is Call Detail Record, which provides logging services via a variety of
+; pluggable backend modules.  Detailed call information can be recorded to
+; databases, files, etc.  Useful for billing, fraud prevention, compliance with
+; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.
+;
+
+[general]
+
+; Define whether or not to use CDR logging.  Setting this to "no" will override
+; any loading of backend CDR modules.  Default is "yes".
+;enable=yes
+
+; Define whether or not to log unanswered calls that don't involve an outgoing
+; party. Setting this to "yes" will make calls to extensions that don't answer
+; and don't set a B side channel (such as by using the Dial application)
+; receive CDR log entries. If this option is set to "no", then those log
+; entries will not be created. Unanswered Calls which get offered to an
+; outgoing line will always receive log entries regardless of this option, and
+; that is the intended behaviour.
+;unanswered = no
+unanswered = yes
+
+; Define whether or not to log congested calls. Setting this to "yes" will
+; report each call that fails to complete due to congestion conditions. Default
+; is "no".
+;congestion = no
+
+; Normally, CDR's are not closed out until after all extensions are finished
+; executing.  By enabling this option, the CDR will be ended before executing
+; the "h" extension and hangup handlers so that CDR values such as "end" and
+; "billsec" may be retrieved inside of of this extension.
+; The default value is "no".
+;endbeforehexten=no
+
+; Normally, the 'billsec' field logged to the backends (text files or databases)
+; is simply the end time (hangup time) minus the answer time in seconds. Internally,
+; asterisk stores the time in terms of microseconds and seconds. By setting
+; initiatedseconds to 'yes', you can force asterisk to report any seconds
+; that were initiated (a sort of round up method). Technically, this is
+; when the microsecond part of the end time is greater than the microsecond
+; part of the answer time, then the billsec time is incremented one second.
+; The default value is "no".
+;initiatedseconds=no
+
+; Define the CDR batch mode, where instead of posting the CDR at the end of
+; every call, the data will be stored in a buffer to help alleviate load on the
+; asterisk server.  Default is "no".
+;
+; WARNING WARNING WARNING
+; Use of batch mode may result in data loss after unsafe asterisk termination
+; ie. software crash, power failure, kill -9, etc.
+; WARNING WARNING WARNING
+;
+;batch=no
+
+; Define the maximum number of CDRs to accumulate in the buffer before posting
+; them to the backend engines.  'batch' must be set to 'yes'.  Default is 100.
+;size=100
+
+; Define the maximum time to accumulate CDRs in the buffer before posting them
+; to the backend engines.  If this time limit is reached, then it will post the
+; records, regardless of the value defined for 'size'.  'batch' must be set to
+; 'yes'.  Note that time is in seconds.  Default is 300 (5 minutes).
+;time=300
+
+; The CDR engine uses the internal asterisk scheduler to determine when to post
+; records.  Posting can either occur inside the scheduler thread, or a new
+; thread can be spawned for the submission of every batch.  For small batches,
+; it might be acceptable to just use the scheduler thread, so set this to "yes".
+; For large batches, say anything over size=10, a new thread is recommended, so
+; set this to "no".  Default is "no".
+;scheduleronly=no
+
+; When shutting down asterisk, you can block until the CDRs are submitted.  If
+; you don't, then data will likely be lost.  You can always check the size of
+; the CDR batch buffer with the CLI "cdr status" command.  To enable blocking on
+; submission of CDR data during asterisk shutdown, set this to "yes".  Default
+; is "yes".
+;safeshutdown=yes
+
+;
+;
+; CHOOSING A CDR "BACKEND"  (what kind of output to generate)
+;
+; To choose a backend, you have to make sure either the right category is
+; defined in this file, or that the appropriate config file exists, and has the
+; proper definitions in it. If there are any problems, usually, the entry will
+; silently ignored, and you get no output.
+;
+; Also, please note that you can generate CDR records in as many formats as you
+; wish. If you configure 5 different CDR formats, then each event will be logged
+; in 5 different places! In the example config files, all formats are commented
+; out except for the cdr-csv format.
+;
+; Here are all the possible back ends:
+;
+;   csv, custom, manager, odbc, pgsql, radius, sqlite, tds
+;    (also, mysql is available via the asterisk-addons, due to licensing
+;     requirements)
+;   (please note, also, that other backends can be created, by creating
+;    a new backend module in the source cdr/ directory!)
+;
+; Some of the modules required to provide these backends will not build or install
+; unless some dependency requirements are met. Examples of this are pgsql, odbc,
+; etc. If you are not getting output as you would expect, the first thing to do
+; is to run the command "make menuselect", and check what modules are available,
+; by looking in the "2. Call Detail Recording" option in the main menu. If your
+; backend is marked with XXX, you know that the "configure" command could not find
+; the required libraries for that option.
+;
+; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
+; file, define the [csv] category in this file. No database necessary. The example
+; config files are set up to provide this kind of output by default.
+;
+; To get custom csv CDR records, make sure the cdr_custom.conf file
+; is present, and contains the proper [mappings] section. The advantage to
+; using this backend, is that you can define which fields to output, and in
+; what order. By default, the example configs are set up to mimic the cdr-csv
+; output. If you don't make any changes to the mappings, you are basically generating
+; the same thing as cdr-csv, but expending more CPU cycles to do so!
+;
+; To get manager events generated, make sure the cdr_manager.conf file exists,
+; and the [general] section is defined, with the single variable 'enabled = yes'.
+;
+; For odbc, make sure all the proper libs are installed, that "make menuselect"
+; shows that the modules are available, and the cdr_odbc.conf file exists, and
+; has a [global] section with the proper variables defined.
+;
+; For pgsql, make sure all the proper libs are installed, that "make menuselect"
+; shows that the modules are available, and the cdr_pgsql.conf file exists, and
+; has a [global] section with the proper variables defined.
+;
+; For logging to radius databases, make sure all the proper libs are installed, that
+; "make menuselect" shows that the modules are available, and the [radius]
+; category is defined in this file, and in that section, make sure the 'radiuscfg'
+; variable is properly pointing to an existing radiusclient.conf file.
+;
+; For logging to sqlite databases, make sure the 'cdr.db' file exists in the log directory,
+; which is usually /var/log/asterisk. Of course, the proper libraries should be available
+; during the 'configure' operation.
+;
+; For tds logging, make sure the proper libraries are available during the 'configure'
+; phase, and that cdr_tds.conf exists and is properly set up with a [global] category.
+;
+; Also, remember, that if you wish to log CDR info to a database, you will have to define
+; a specific table in that databse to make things work! See the doc directory for more details
+; on how to create this table in each database.
+;
+
+[csv]
+usegmtime=yes    ; log date/time in GMT.  Default is "no"
+loguniqueid=yes  ; log uniqueid.  Default is "no"
+loguserfield=yes ; log user field.  Default is "no"
+accountlogs=yes  ; create separate log file for each account code. Default is "yes"
+newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns (peeraccount, linkedid, sequence).
+                   ; Default is "no".
+
+;[radius]
+;usegmtime=yes    ; log date/time in GMT
+;loguniqueid=yes  ; log uniqueid
+;loguserfield=yes ; log user field
+; Set this to the location of the radiusclient-ng configuration file
+; The default is /etc/radiusclient-ng/radiusclient.conf
+;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf

version_1.8.2/pabx/etc/asterisk/cdr_adaptive_odbc.conf

@@ -0,0 +1,59 @@
+; The point of this module is to allow you log whatever you like in terms of
+; the CDR variables.  Do you want to log uniqueid?  Then simply ensure that
+; your table has that column.  If you don't want the column, ensure that it
+; does not exist in the table structure.  If you'd like to call uniqueid
+; something else in your table, simply provide an alias in the configuration
+; file that maps the standard CDR field name (uniqueid) to whatever column
+; name you like.  Perhaps you'd like some extra CDR values logged that aren't
+; in the standard repertoire of CDR variables (some that come to mind are
+; certain values used for LCR:  route, per_minute_cost, and per_minute_price).
+; Simply set those CDR variables in your dialplan, i.e. Set(CDR(route)=27),
+; ensure that a corresponding column exists in your table, and cdr_adaptive_odbc
+; will do the rest.
+;
+; This configuration defines the connections and tables for which CDRs may
+; be populated.  Each context specifies a different CDR table to be used.
+;
+; The columns in the tables should match up word-for-word (case-insensitive)
+; to the CDR variables set in the dialplan.  The natural advantage to this
+; system is that beyond setting up the configuration file to tell you what
+; tables to look at, there isn't anything more to do beyond creating the
+; columns for the fields that you want, and populating the corresponding
+; CDR variables in the dialplan.  For the builtin variables only, you may
+; create aliases for the real column name.
+;
+; Please note that after adding columns to the database, it is necessary to
+; reload this module to get the new column names and types read.
+;
+; Warning: if you specify two contexts with exactly the same connection and
+; table names, you will get duplicate records in that table.  So be careful.
+;
+
+;[first]
+;connection=mysql1
+;table=cdr
+
+;[second]
+;connection=mysql1
+;table=extracdr
+
+;[third]
+;connection=sqlserver
+;table=AsteriskCDR
+;schema=public ; for databases which support schemas
+;usegmtime=yes ; defaults to no
+;alias src => source
+;alias channel => source_channel
+;alias dst => dest
+;alias dstchannel => dest_channel
+;
+; Any filter specified MUST match exactly or the CDR will be discarded
+;filter accountcode => somename
+;filter src => 123
+; Negative filters are also now available
+;filter src != 456
+;
+; Additionally, we now support setting static values per column.  The reason
+; for this is to allow different sections to specify different values for
+; a certain named column, presumably separated by filters.
+;static "Some Special Value" => identifier_code

version_1.8.2/pabx/etc/asterisk/cdr_custom.conf

@@ -0,0 +1,16 @@
+;
+; Mappings for custom config file
+;
+; To get your CSV output in a format tailored to your liking, uncomment the
+; following lines and look for the output in the cdr-custom directory (usually
+; in /var/log/asterisk).  Depending on which mapping you uncomment, you may see
+; Master.csv, Simple.csv, or both.
+;
+[mappings]
+Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)},${CSV_QUOTE(${CDR(direcao)})},${CSV_QUOTE(${CDR(ramal_origem)})},${CSV_QUOTE(${CDR(fora_horario)})}
+
+
+;
+; High Resolution Time for billsec and duration fields
+;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
+;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}

version_1.8.2/pabx/etc/asterisk/cdr_manager.conf

@@ -0,0 +1,6 @@
+;
+; Asterisk Call Management CDR
+;
+[general]
+enabled = yes
+

version_1.8.2/pabx/etc/asterisk/cdr_mysql.conf

@@ -0,0 +1,62 @@
+;
+; Note - if the database server is hosted on the same machine as the
+; asterisk server, you can achieve a local Unix socket connection by
+; setting hostname=localhost
+;
+; port and sock are both optional parameters.  If hostname is specified
+; and is not "localhost" (you can use address 127.0.0.1 instead), then
+; cdr_mysql will attempt to connect to the port specified or use the
+; default port.  If hostname is not specified or if hostname is
+; "localhost", then cdr_mysql will attempt to connect to the socket file
+; specified by sock or otherwise use the default socket file.
+;
+;[global]
+;hostname=database.host.name
+;dbname=asteriskcdrdb
+;table=cdr
+;password=password
+;user=asteriskcdruser
+;port=3306
+;sock=/tmp/mysql.sock
+; By default CDRs are logged in the system's time zone
+;cdrzone=UTC               ; log CDRs with UTC
+;usegmtime=yes ;log date/time in GMT.  Default is "no"
+;cdrzone=America/New_York  ; or use a specific time zone
+;
+; If your system's locale differs from mysql database character set,
+; cdr_mysql can damage non-latin characters in CDR variables. Use this
+; option to protect your data.
+;charset=koi8r
+;
+; Older versions of cdr_mysql set the calldate field to whenever the
+; record was posted, rather than the start date of the call.  This flag
+; reverts to the old (incorrect) behavior.  Note that you'll also need
+; to comment out the "start=calldate" alias, below, to use this.
+;compat=no
+;
+; ssl connections (optional)
+;ssl_ca=<path to CA cert>
+;ssl_cert=<path to cert>
+;ssl_key=<path to keyfile>
+;
+; You may also configure the field names used in the CDR table.
+;
+[columns]
+;static "<value>" => <column>
+;alias <cdrvar> => <column>
+alias start => calldate
+;alias clid => <a_field_not_named_clid>
+;alias src => <a_field_not_named_src>
+;alias dst => <a_field_not_named_dst>
+;alias dcontext => <a_field_not_named_dcontext>
+;alias channel => <a_field_not_named_channel>
+;alias dstchannel => <a_field_not_named_dstchannel>
+;alias lastapp => <a_field_not_named_lastapp>
+;alias lastdata => <a_field_not_named_lastdata>
+;alias duration => <a_field_not_named_duration>
+;alias billsec => <a_field_not_named_billsec>
+;alias disposition => <a_field_not_named_disposition>
+;alias amaflags => <a_field_not_named_amaflags>
+;alias accountcode => <a_field_not_named_accountcode>
+;alias userfield => <a_field_not_named_userfield>
+;alias uniqueid => <a_field_not_named_uniqueid>

version_1.8.2/pabx/etc/asterisk/cdr_odbc.conf

@@ -0,0 +1,12 @@
+;
+; cdr_odbc.conf
+;
+
+;[global]
+;dsn=MySQL-test
+;loguniqueid=yes
+;dispositionstring=yes
+;table=cdr		;"cdr" is default table name
+;usegmtime=no             ; set to "yes" to log in GMT
+;hrtime=yes		;Enables microsecond accuracy with the billsec and duration fields
+;newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns (peeraccount, linkedid, sequence)

version_1.8.2/pabx/etc/asterisk/cdr_pgsql.conf

@@ -0,0 +1,8 @@
+[global]
+hostname=postgres
+port=5432
+dbname=pbx
+password=ctepgSQL
+user=contacte
+table=ast_bilhetes
+encoding=LATIN1

version_1.8.2/pabx/etc/asterisk/cdr_sqlite3_custom.conf

@@ -0,0 +1,10 @@
+;
+; Mappings for custom config file
+;
+[master] ; currently, only file "master.db" is supported, with only one table at a time.
+;table => cdr
+;columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield, test
+;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}','${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'
+
+;Enable High Resolution Times for billsec and duration fields
+;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration,f)}','${CDR(billsec,f)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'

version_1.8.2/pabx/etc/asterisk/cdr_syslog.conf

@@ -0,0 +1,83 @@
+;
+; Asterisk Call Detail Records (CDR) - Syslog Backend
+;
+
+; The cdr_syslog module writes CDRs using the facilities provided by syslog.
+;
+; Not only must you configure cdr_syslog from this file (cdr_syslog.conf) but
+; you will also need to make changes to your /etc/syslog.conf before CDRs will
+; be written to syslog.
+;
+; As an example, you can add the following to /etc/syslog.conf:
+;
+;    local4.info        /var/log/asterisk-cdr.log
+;
+; And then instruct syslogd to re-read the configuration file by sending it a
+; HUP signal.  On Linux this can be done like this:
+;
+;    kill -HUP `cat /var/run/syslogd.pid`
+;
+; Finally, you will need to uncomment the [cdr-simple] section below, and restart
+; Asterisk.  When calls are placed, you should start seeing records appear in
+; /var/log/asterisk-cdr.log.
+
+[general]
+; Facility
+;
+; The 'facility' keyword specifies the syslog facility to use when writing out
+; CDRs.
+;
+; Accepted values: One of the following:
+;                  user, local0, local1, local2, local3, local4, local5, local6
+;                  and local7.
+;
+;                  Note: Depending on your platform, the following may also be
+;                        available:
+;                        auth, authpriv, cron, daemon, ftp, kern, lpr, mail,
+;                        news, syslog, and uucp.
+;
+; Default value:   local4
+
+;facility=local0
+
+; Priority
+;
+; Use the 'priority' keyword to select which of the syslog priority levels to
+; use when logging CDRs.
+;
+; Accepted values: One of the following:
+;                  alert, crit, debug, emerg, err, info, notice, warning
+; Default value:   info
+
+;priority=warn
+
+; Note: The settings for 'facility' and 'priority' in the [general] section
+;       define the default values for all of the logging locations created
+;       below in separate sections.
+
+;[cdr-master]
+;facility = local5
+;priority = debug
+
+; Template
+;
+; The 'template' value allows you to specify a custom format for messages
+; written to syslog.  This is similar to how cdr_custom is configured.
+;
+; Allowed values: A diaplan style string.
+; Default value:  None, this is required field.
+;
+; Note: Because of the way substitution is done, the only meaningful values
+;       available when the record is logged are those available via the CDR()
+;       dialplan function.  All other channel variables will be unavailable.
+
+;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
+
+; High Resolution Time for billsec and duration fields
+;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration,f)}","${CDR(billsec,f)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
+;[cdr-simple]
+
+; Since we don't specify a facility or priority for this logging location, the
+; records will use the defaults specified in the [general] section.
+
+;template = "We received a call from ${CDR(src)}"

version_1.8.2/pabx/etc/asterisk/cdr_tds.conf

@@ -0,0 +1,77 @@
+;
+; Asterisk Call Detail Records (CDR) - FreeTDS Backend
+;
+
+;[global]
+
+; Connection
+;
+; Use the 'connection' keyword to specify one of the instance names from your
+; 'freetds.conf' file.  Note that 'freetds.conf' is not an Asterisk
+; configuration file, but one specific to the FreeTDS library.  See the FreeTDS
+; documentation on 'freetds.conf' for more information:
+;
+;         http://www.freetds.org/userguide/freetdsconf.htm
+;
+; Accepted values: One of the connections specified in freetds.conf
+
+;connection=ConnectionFromFreeTDSConf
+
+; Database Name
+;
+; The 'dbname' keyword specifies the database name to use when logging CDRs.
+;
+; Accepted values: Any valid database name
+
+;dbname=AsteriskCDRs
+
+; Database Table Name
+;
+; The 'table' keyword identifies which database table is used to log CDRs.
+;
+; Accepted value: Any valid table name
+; Default value:  If not specified, a table named 'cdr' is assumed
+
+;table=cdr
+
+; Credentials
+;
+; The 'username' and 'password' keywords specify the user credentials that
+; Asterisk should use when connecting to the database.
+;
+; Accepted value: Any valid username and password
+
+;username=mangUsr
+;password=
+
+; Language
+;
+; The 'language' keyword changes the language which are used for error and
+; information messages returned by SQL Server.  Each database and user has their
+; own default value, and this default can be overriden here.
+;
+; Accepted value: Any language installed on the target SQL Server.
+; Default value:  us_english
+
+;language=us_english
+
+; Character Set
+;
+; The 'charset' setting is used to change the character set used when connecting
+; to the database server.  Each database and database user has their own
+; character set setting, and this default can be overriden here.
+;
+; Accepted value: Any valid character set available on the target SQL server.
+; Default value:  iso_1
+
+;charset=BIG5
+
+; High Resolution Times
+;
+; The 'hrtime' setting is used to store high resolution (sub second) times for
+; billsec and duration fields.
+;
+; Accepted value: true or false
+; Default value: false
+
+;hrtime=false

version_1.8.2/pabx/etc/asterisk/cel.conf

@@ -0,0 +1,116 @@
+;
+; Asterisk Channel Event Logging (CEL)
+;
+
+; Channel Event Logging is a mechanism to provide fine-grained event information
+; that can be used to generate billing information. Such event information can
+; be recorded to various backend modules.
+;
+
+[general]
+
+; CEL Activation
+;
+; Use the 'enable' keyword to turn CEL on or off.
+;
+; Accepted values: yes and no
+; Default value:   no
+
+enable=no
+
+; Application Tracking
+;
+; Use the 'apps' keyword to specify the list of applications for which you want
+; to receive CEL events.  This is a comma separated list of Asterisk dialplan
+; applications, such as Dial, Queue, and Park.
+;
+; Accepted values: A comma separated list of Asterisk dialplan applications
+; Default value:   none
+;
+; Note: You may also use 'all' which will result in CEL events being reported
+;       for all Asterisk applications.  This may affect Asterisk's performance
+;       significantly.
+
+apps=dial,park
+
+; Event Tracking
+;
+; Use the 'events' keyword to specify the list of events which you want to be
+; raised when they occur.  This is a comma separated list of the values in the
+; table below.
+;
+; Accepted values: A comma separated list of one or more of the following:
+;  ALL              -- Generate entries on all events
+;  CHAN_START       -- The time a channel was created
+;  CHAN_END         -- The time a channel was terminated
+;  ANSWER           -- The time a channel was answered (ie, phone taken off-hook)
+;  HANGUP           -- The time at which a hangup occurred
+;  BRIDGE_ENTER       -- The time a channel was connected into a conference room
+;  BRIDGE_EXIT        -- The time a channel was removed from a conference room
+;  APP_START        -- The time a tracked application was started
+;  APP_END          -- the time a tracked application ended
+;  PARK_START       -- The time a call was parked
+;  PARK_END         -- Unpark event
+;  BLINDTRANSFER    -- When a blind transfer is initiated
+;  ATTENDEDTRANSFER -- When an attended transfer is initiated
+;  PICKUP           -- This channel picked up the specified channel
+;  FORWARD          -- This channel is being forwarded somewhere else
+;  LINKEDID_END     -- The last channel with the given linkedid is retired
+;  USER_DEFINED     -- Triggered from the dialplan, and has a name given by the
+;                      user
+;  LOCAL_OPTIMIZE   -- A local channel pair is optimizing away.
+;
+; Default value: none
+;                (Track no events)
+
+events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT
+
+; Date Format
+;
+; Use the 'dateformat' keyword to specify the date format used when CEL events
+; are raised.
+;
+; Accepted values: A strftime format string (see man strftime)
+;
+; Example: "%F %T"
+;  -> This gives the date and time in the format "2009-06-23 17:02:35"
+;
+; If this option is not specified, the default format is "<seconds>.<microseconds>"
+; since epoch.  The microseconds field will always be 6 digits in length, meaning it
+; may have leading zeros.
+;
+;dateformat = %F %T
+
+;
+; Asterisk Manager Interface (AMI) CEL Backend
+;
+[manager]
+
+; AMI Backend Activation
+;
+; Use the 'enable' keyword to turn CEL logging to the Asterisk Manager Interface
+; on or off.
+;
+; Accepted values: yes and no
+; Default value:   no
+;enabled=yes
+
+; Use 'show_user_defined' to put "USER_DEFINED" in the EventName header,
+; instead of (by default) just putting the user defined event name there.
+; When enabled the UserDefType header is added for user defined events to
+; provide the user defined event name.
+;
+;show_user_defined=yes
+
+;
+; RADIUS CEL Backend
+;
+[radius]
+;
+; Log date/time in GMT
+;usegmtime=yes
+;
+; Set this to the location of the radiusclient-ng configuration file
+; The default is /etc/radiusclient-ng/radiusclient.conf
+;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
+;

version_1.8.2/pabx/etc/asterisk/cel_custom.conf

@@ -0,0 +1,39 @@
+;
+; Asterisk Channel Event Logging (CEL) - Custom CSV Backend
+;
+
+; This is the configuration file for the customizable CSV backend for CEL
+; logging.
+;
+; In order to create custom CSV logs for CEL, uncomment the template below
+; (Master.csv) and start Asterisk.  Once CEL events are generated, a file will
+; appear in the following location:
+;
+;                 /var/log/asterisk/cel-custom/Master.csv
+;
+; (Note that /var/log/asterisk is the default and may differ on your system)
+;
+; You can also create more than one template if desired.  All logs will appear
+; in the cel-custom directory under your Asterisk logs directory.
+;
+
+;
+; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
+; details from the CEL event.  There are also a few variables created by this
+; module that can be used in a mapping:
+;
+;    eventtype   - The name of the CEL event.
+;    eventtime   - The timestamp of the CEL event.
+;    eventenum   - Like eventtype but is "USER_DEFINED" for a user defined event.
+;    userdeftype - User defined event type name from CELGenUserEvent().
+;    eventextra  - Extra data included with this CEL event, typically along with
+;                  an event of type USER_DEFINED from CELGenUserEvent().
+;    BRIDGEPEER  - Bridged peer channel name at the time of the CEL event.
+;                  CHANNEL(peer) could also be used.
+;
+;[mappings]
+;Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE(${eventtime})},${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE(${CALLERID(num)})},${CSV_QUOTE(${CALLERID(ANI)})},${CSV_QUOTE(${CALLERID(RDNIS)})},${CSV_QUOTE(${CALLERID(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},${CSV_QUOTE(${CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},${CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL(appdata)})},${CSV_QUOTE(${CHANNEL(amaflags)})},${CSV_QUOTE(${CHANNEL(accountcode)})},${CSV_QUOTE(${CHANNEL(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},${CSV_QUOTE(${BRIDGEPEER})},${CSV_QUOTE(${CHANNEL(userfield)})},${CSV_QUOTE(${userdeftype})},${CSV_QUOTE(${eventextra})}
+
+
+[mappings]
+Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE(${eventtime})},${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE(${CALLERID(num)})},${CSV_QUOTE(${CALLERID(ANI)})},${CSV_QUOTE(${CALLERID(RDNIS)})},${CSV_QUOTE(${CALLERID(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},${CSV_QUOTE(${CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},${CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL(appdata)})},${CSV_QUOTE(${CHANNEL(amaflags)})},${CSV_QUOTE(${CHANNEL(accountcode)})},${CSV_QUOTE(${CHANNEL(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},${CSV_QUOTE(${BRIDGEPEER})},${CSV_QUOTE(${CHANNEL(userfield)})},${CSV_QUOTE(${userdeftype})},${CSV_QUOTE(${eventextra})}

version_1.8.2/pabx/etc/asterisk/cel_odbc.conf

@@ -0,0 +1,108 @@
+;
+; Asterisk Channel Event Logging (CEL) - Adaptive ODBC Backend
+;
+
+; General module options category.
+[general]
+; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
+; instead of (by default) just putting the user defined event name there.
+;
+;show_user_defined=yes
+
+; This configuration defines the connections and tables for which CEL records
+; may be populated.  Each context specifies a different CEL table to be used.
+;
+; The columns in the tables should match up word-for-word (case-insensitive) to
+; the CEL variables set in the dialplan.  The natural advantage to this system
+; is that beyond setting up the configuration file to tell you what tables to
+; look at, there isn't anything more to do beyond creating the columns for the
+; fields that you want, and populating the corresponding CEL variables in the
+; dialplan.
+;
+; Please note that after adding columns to the database, it is necessary to
+; reload this module to get the new column names and types read.
+;
+; Warning: if you specify two contexts with exactly the same connection and
+; table names, you will get duplicate records in that table.  So be careful.
+;
+; CEL FIELDS:
+;	eventtype
+;	  CHANNEL_START = 1
+;	  CHANNEL_END = 2
+;	  HANGUP = 3
+;	  ANSWER = 4
+;	  APP_START = 5
+;	  APP_END = 6
+;	  BRIDGE_START = 7
+;	  BRIDGE_END = 8
+;	  CONF_START = 9
+;	  CONF_END = 10
+;	  PARK_START = 11
+;	  PARK_END = 12
+;	  BLINDTRANSFER = 13
+;	  ATTENDEDTRANSFER = 14
+;	  TRANSFER = 15
+;	  HOOKFLASH = 16
+;	  3WAY_START = 17
+;	  3WAY_END = 18
+;	  CONF_ENTER = 19
+;	  CONF_EXIT = 20
+;	  USER_DEFINED = 21
+;	  LINKEDID_END = 22
+;	  BRIDGE_UPDATE = 23
+;	  PICKUP = 24
+;	  FORWARD = 25
+;	eventtime  (timeval, includes microseconds)
+;	userdeftype (set only if eventtype == USER_DEFINED)
+;	cid_name
+;	cid_num
+;	cid_ani
+;	cid_rdnis
+;	cid_dnid
+;	exten
+;	context
+;	channame
+;	appname
+;	appdata
+;	accountcode
+;	peeraccount
+;	uniqueid
+;	linkedid
+;	amaflags  (an int)
+;	userfield
+;	peer
+;	extra
+
+; The point of this module is to allow you log whatever you like in terms of the
+; CEL variables.  Do you want to log uniqueid?  Then simply ensure that your
+; table has that column.  If you don't want the column, ensure that it does not
+; exist in the table structure.  If you'd like to call uniqueid something else
+; in your table, simply provide an alias in this file that maps the standard CEL
+; field name (uniqueid) to whatever column name you like.
+
+;[first]
+;connection=mysql1
+;table=cel
+
+;[second]
+;connection=mysql1
+;table=extracel
+
+;[third]
+;connection=sqlserver
+;table=AsteriskCEL
+;usegmtime=yes ; defaults to no
+;allowleapsecond=no ; allow leap second in SQL column for eventtime, default yes.
+;alias src => source
+;alias channel => source_channel
+;alias dst => dest
+;alias dstchannel => dest_channel
+
+; Any filter specified MUST match exactly or the event will be discarded
+;filter accountcode => somename
+;filter src => 123
+
+; Additionally, we now support setting static values per column.  Reason
+; for this is to allow different sections to specify different values for
+; a certain named column, presumably separated by filters.
+;static "Some Special Value" => identifier_code

version_1.8.2/pabx/etc/asterisk/cel_pgsql.conf

@@ -0,0 +1,93 @@
+;
+; Asterisk Channel Event Logging (CEL) - PostgreSQL Backend
+;
+
+; Sample Asterisk config file for CEL logging to PostgreSQL
+;
+; CEL field names:
+;
+;	eventtype
+;	  CHANNEL_START = 1
+;	  CHANNEL_END = 2
+;	  HANGUP = 3
+;	  ANSWER = 4
+;	  APP_START = 5
+;	  APP_END = 6
+;	  BRIDGE_START = 7
+;	  BRIDGE_END = 8
+;	  CONF_START = 9
+;	  CONF_END = 10
+;	  PARK_START = 11
+;	  PARK_END = 12
+;	  BLINDTRANSFER = 13
+;	  ATTENDEDTRANSFER = 14
+;	  TRANSFER = 15
+;	  HOOKFLASH = 16
+;	  3WAY_START = 17
+;	  3WAY_END = 18
+;	  CONF_ENTER = 19
+;	  CONF_EXIT = 20
+;	  USER_DEFINED = 21
+;	  LINKEDID_END = 22
+;	  BRIDGE_UPDATE = 23
+;	  PICKUP = 24
+;	  FORWARD = 25
+;	eventtime  (timeval, includes microseconds)
+;	userdeftype (set only if eventtype == USER_DEFINED)
+;	cid_name
+;	cid_num
+;	cid_ani
+;	cid_rdnis
+;	cid_dnid
+;	exten
+;	context
+;	channame
+;	appname
+;	appdata
+;	accountcode
+;	peeraccount
+;	uniqueid
+;	linkedid
+;	amaflags  (an int)
+;	userfield
+;	peer
+;	extra
+
+;CREATE TABLE pbx_cel (
+;        id bigserial , 
+;        eventtype varchar (32) NOT NULL ,
+;        eventtime timestamp NOT NULL ,
+;        userdeftype varchar(256) NOT NULL ,
+;        cid_name varchar (128) NOT NULL , 
+;        cid_num varchar (128) NOT NULL ,
+;        cid_ani varchar (128) NOT NULL , 
+;        cid_rdnis varchar (128) NOT NULL ,
+;        cid_dnid varchar (128) NOT NULL ,
+;        exten varchar (128) NOT NULL ,
+;        context varchar (128) NOT NULL , 
+;        channame varchar (128) NOT NULL ,
+;        appname varchar (128) NOT NULL ,
+;        appdata varchar (128) NOT NULL , 
+;        amaflags int NOT NULL ,
+;        accountcode varchar (32) NOT NULL ,
+;        peeraccount varchar (32) NOT NULL ,
+;        uniqueid varchar (160) NOT NULL ,
+;        linkedid varchar (160) NOT NULL , 
+;        userfield varchar (256) NOT NULL ,
+;        peer varchar (128) NOT NULL 
+;);
+
+
+[global]
+; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
+; instead of (by default) just putting the user defined event name there.
+;
+;show_user_defined=yes
+
+hostname=127.0.0.1
+port=5432
+dbname=pbx
+password=ctepgSQL
+user=contacte
+table=pbx_cel		;SQL table where CEL's will be inserted
+;appname=asterisk   ; Postgres application_name support (optional). Whitespace not allowed.

version_1.8.2/pabx/etc/asterisk/cel_sqlite3_custom.conf

@@ -0,0 +1,24 @@
+;
+; Asterisk Channel Event Logging (CEL) - SQLite 3 Backend
+;
+
+;
+; Mappings for sqlite3 config file
+;
+; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
+; details from the CEL event.  There are also a few variables created by this
+; module that can be used in a mapping:
+;
+;    eventtype   - The name of the CEL event.
+;    eventtime   - The timestamp of the CEL event.
+;    eventenum   - Like eventtype but is "USER_DEFINED" for a user defined event.
+;    userdeftype - User defined event type name from CELGenUserEvent().
+;    eventextra  - Extra data included with this CEL event, typically along with
+;                  an event of type USER_DEFINED from CELGenUserEvent().
+;    BRIDGEPEER  - Bridged peer channel name at the time of the CEL event.
+;                  CHANNEL(peer) could also be used.
+;
+;[master] ; currently, only file "master.db" is supported, with only one table at a time.
+;table	=> cel
+;columns	=> eventtype, eventtime, cidname, cidnum, cidani, cidrdnis, ciddnid, context, exten, channame, appname, appdata, amaflags, accountcode, uniqueid, userfield, peer, userdeftype, eventextra
+;values	=> '${eventtype}','${eventtime}','${CALLERID(name)}','${CALLERID(num)}','${CALLERID(ANI)}','${CALLERID(RDNIS)}','${CALLERID(DNID)}','${CHANNEL(context)}','${CHANNEL(exten)}','${CHANNEL(channame)}','${CHANNEL(appname)}','${CHANNEL(appdata)}','${CHANNEL(amaflags)}','${CHANNEL(accountcode)}','${CHANNEL(uniqueid)}','${CHANNEL(userfield)}','${BRIDGEPEER}','${userdeftype}','${eventextra}'

version_1.8.2/pabx/etc/asterisk/cel_tds.conf

@@ -0,0 +1,69 @@
+;
+; Asterisk Channel Event Logging (CEL) - FreeTDS Backend
+;
+
+;[global]
+
+; Connection
+;
+; Use the 'connection' keyword to specify one of the instance names from your
+; 'freetds.conf' file.  Note that 'freetds.conf' is not an Asterisk
+; configuration file, but one specific to the FreeTDS library.  See the FreeTDS
+; documentation on 'freetds.conf' for more information:
+;
+;         http://www.freetds.org/userguide/freetdsconf.htm
+;
+; Accepted values: One of the connections specified in freetds.conf
+
+;connection=ConnectionFromFreeTDSConf
+
+; Database Name
+;
+; The 'dbname' keyword specifies the database name to use when logging CEL
+; records.
+;
+; Accepted values: Any valid database name
+
+;dbname=MalicoHN
+
+; Database Table Name
+;
+; The 'table' keyword identifies which database table is used to log CEL
+; records.
+;
+; Accepted value: Any valid table name
+; Default value:  If not specified, a table named 'cel' is assumed
+
+;table=cel
+
+; Credentials
+;
+; The 'username' and 'password' keywords specify the user credentials that
+; Asterisk should use when connecting to the database.
+;
+; Accepted value: Any valid username and password
+
+;username=mangUsr
+;password=
+
+; Language
+;
+; The 'language' keyword changes the language which are used for error and
+; information messages returned by SQL Server.  Each database and user has their
+; own default value, and this default can be overriden here.
+;
+; Accepted value: Any language installed on the target SQL Server.
+; Default value:  Server default
+
+;language=us_english
+
+; Character Set
+;
+; The 'charset' setting is used to change the character set used when connecting
+; to the database server.  Each database and database user has their own
+; character set setting, and this default can be overriden here.
+;
+; Accepted value: Any valid character set available on the target server.
+; Default value:  Server setting
+
+;charset=BIG5

version_1.8.2/pabx/etc/asterisk/chan_dahdi.conf

@@ -0,0 +1,7 @@
+[trunkgroups]
+
+[channels]
+#include chan_dahdi_troncos.conf
+#include chan_dahdi_troncos_general.conf
+#include chan_dahdi_ramais_general.conf
+#include chan_dahdi_ramais.conf

version_1.8.2/pabx/etc/asterisk/chan_dahdi_ramais_general.conf

@@ -0,0 +1,21 @@
+;Config. Portas FXS
+language=pt_br
+usecallerid=yes
+callwaiting=no
+usecallingpres=yes
+callwaitingcallerid=yes
+threewaycalling=yes
+transfer=yes
+canpark=yes
+cancallforward=yes
+callreturn=yes
+;rxgain=3
+;txgain=-7
+rxgain=0
+txgain=-6
+echocancel=128
+echocancelwhenbridged=yes
+echotraining=yes
+relaxdtmf=yes
+rxflash=600
+group=10

version_1.8.2/pabx/etc/asterisk/chan_dahdi_troncos_general.conf

@@ -0,0 +1,15 @@
+language=pt_br
+usecallerid=yes
+callwaiting=no
+usecallingpres=yes
+callwaitingcallerid=yes
+;rxgain=1
+;txgain=-7
+;rxgain=4
+;txgain=4
+;echocancel=128
+;echocancelwhenbridged=yes
+;echotraining=yes
+;busydetect=yes
+;busycount=4
+relaxdtmf=yes

version_1.8.2/pabx/etc/asterisk/chan_mobile.conf

@@ -0,0 +1,69 @@
+;
+; chan_mobile.conf
+; configuration file for chan_mobile
+;
+
+[general]
+interval=30		; Number of seconds between trying to connect to devices.
+
+; The following is a list of adapters we use.
+; id must be unique and address is the bdaddr of the adapter from hciconfig.
+; Each adapter may only have one device (headset or phone) connected at a time.
+; Add an [adapter] entry for each adapter you have.
+
+[adapter]
+id=blue
+address=00:09:DD:60:01:A3
+;forcemaster=yes	; attempt to force adapter into master mode. default is no.
+;alignmentdetection=yes ; enable this if you sometimes get 'white noise' on asterisk side of the call
+			; its a bug in the bluetooth adapter firmware, enabling this will compensate for it.
+			; default is no.
+
+[adapter]
+id=dlink
+address=00:80:C8:35:52:78
+
+; The following is a list of the devices we deal with.
+; Every device listed below will be available for calls in and out of Asterisk.
+; Each device needs an adapter=xxxx entry which determines which bluetooth adapter is used.
+; Use the CLI command 'mobile search' to discover devices.
+; Use the CLI command 'mobile show devices' to see device status.
+;
+; To place a call out through a mobile phone use Dial(Mobile/[device]/NNN.....) or Dial(Mobile/gn/NNN......) in your dialplan.
+; To call a headset use Dial(Mobile/[device]).
+
+[LGTU550]
+address=00:E0:91:7F:46:44	; the address of the phone
+port=4				; the rfcomm port number (from mobile search)
+context=incoming-mobile		; dialplan context for incoming calls
+adapter=dlink			; adapter to use
+group=1				; this phone is in channel group 1
+;sms=no				; support SMS, defaults to yes
+;nocallsetup=yes		; set this only if your phone reports that it supports call progress notification, but does not do it. Motorola L6 for example.
+
+[blackberry]
+address=00:60:57:32:7E:B2
+port=2
+context=incoming-mobile
+adapter=dlink
+group=1
+;blackberry=yes			; set this if you are using a blackberry device
+
+[6310i]
+address=00:60:57:32:7E:B1
+port=13
+context=incoming-mobile
+adapter=dlink
+group=1				; this phone is in channel group 1 also.
+
+[headset]
+address=00:0B:9E:11:AE:C6
+port=1
+type=headset			; This is a headset, not a Phone !
+adapter=blue
+
+[headset1]
+address=00:0B:9E:11:74:A5
+port=1
+type=headset
+adapter=dlink

version_1.8.2/pabx/etc/asterisk/cli.conf

@@ -0,0 +1,12 @@
+;
+; Asterisk CLI configuration
+;
+
+[startup_commands]
+;
+; Any commands listed in this section will get automatically executed
+; when Asterisk starts as a daemon or foreground process (-c).
+;
+;sip set debug on = yes
+;core set verbose 3 = yes
+;core set debug 1 = yes

version_1.8.2/pabx/etc/asterisk/cli_aliases.conf

@@ -0,0 +1,203 @@
+;
+; CLI Aliases configuration
+;
+; This module also registers a "cli show aliases" CLI command to list
+; configured CLI aliases.
+
+[general]
+; Here you define what alias templates you want to use. You can also define
+; multiple templates to use as well. If you do, and there is a conflict, then
+; the first alias defined will win.
+;
+template = friendly		; By default, include friendly aliases
+;template = asterisk_1dot2	; Asterisk 1.2 style syntax
+;template = asterisk_1dot4	; Asterisk 1.4 style syntax
+;template = individual_custom	; see [individual_custom] example below which
+				; includes a list of aliases from an external
+				; file
+
+
+; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
+; sometimes we run into an issue between being consistant with this format
+; in the core system, and maintaining system friendliness. In order to get
+; around this we're providing some useful aliases by default.
+;
+[friendly]
+hangup request=channel request hangup
+originate=channel originate
+help=core show help
+pri intense debug span=pri set debug intense span
+reload=module reload
+pjsip reload=module reload res_pjsip.so res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so
+
+; CLI Alias Templates
+; -------------------
+;
+; You can define several alias templates.
+; It works with context templates like all other configuration files
+;
+;[asterisk](!)
+; To create an alias you simply set the variable name as the alias and variable
+; value as the real CLI command you want executed
+;
+;die die die=stop now
+
+;[asterisk_1dot6](asterisk)
+; Alias for making voicemail reload actually do module reload app_voicemail.so
+;voicemail reload=module reload app_voicemail.so
+; This will make the CLI command "mr" behave as though it is "module reload".
+;mr=module reload
+;
+;
+; In addition, you could also include a flat file of aliases which is loaded by
+; the [individual_custom] template in the [general] section.
+;
+;[individual_custom]
+;#include "/etc/asterisk/aliases"
+
+; Implemented CLI Alias Templates
+; -------------------------------
+;
+; Below here we have provided you with some templates, easily allowing you to
+; utilize previous Asterisk CLI commands with any version of Asterisk. In this
+; way you will be able to use Asterisk 1.2 and 1.4 style CLI syntax with any
+; version Asterisk going forward into the future.
+;
+; We have also separated out the vanilla syntax into a context template which
+; allows you to keep your custom changes separate of the standard templates
+; we have provided you. In this way you can clearly see your custom changes,
+; and also allowing you to combine various templates as you see fit.
+;
+; The naming scheme we have used is recommended, but certainly is not enforced
+; by Asterisk. If you wish to use the provided templates, simply define the
+; context name which does not utilize the '_tpl' at the end. For example,
+; if you would like to use the Asterisk 1.2 style syntax, define in the
+; [general] section
+
+[asterisk_1dot2_tpl](!)
+show channeltypes=core show channeltypes
+show channeltype=core show channeltype
+show manager command=manager show command
+show manager commands=manager show commands
+show manager connected=manager show connected
+show manager eventq=manager show eventq
+rtp no debug=rtp set debug off
+rtp rtcp debug ip=rtcp debug ip
+rtp rtcp debug=rtcp debug
+rtp rtcp no debug=rtcp debug off
+rtp rtcp stats=rtcp stats
+rtp rtcp no stats=rtcp stats off
+stun no debug=stun debug off
+udptl no debug=udptl debug off
+show image formats=core show image formats
+show file formats=core show file formats
+show applications=core show applications
+show functions=core show functions
+show switches=core show switches
+show hints=core show hints
+show globals=core show globals
+show function=core show function
+show application=core show application
+set global=core set global
+show dialplan=dialplan show
+show codecs=core show codecs
+show audio codecs=core show audio codecs
+show video codecs=core show video codecs
+show image codecs=core show image codecs
+show codec=core show codec
+moh classes show=moh show classes
+moh files show=moh show files
+agi no debug=agi debug off
+show agi=agi show
+dump agihtml=agi dumphtml
+show features=feature show
+show indications=indication show
+answer=console answer
+hangup=console hangup
+flash=console flash
+dial=console dial
+mute=console mute
+unmute=console unmute
+transfer=console transfer
+send text=console send text
+autoanswer=console autoanswer
+oss boost=console boost
+console=console active
+save dialplan=dialplan save
+add extension=dialplan add extension
+remove extension=dialplan remove extension
+add ignorepat=dialplan add ignorepat
+remove ignorepat=dialplan remove ignorepat
+include context=dialplan add include
+dont include=dialplan remove include
+extensions reload=dialplan reload
+show translation=core show translation
+convert=file convert
+show queue=queue show
+add queue member=queue add member
+remove queue member=queue remove member
+ael no debug=ael nodebug
+sip debug=sip set debug
+sip no debug=sip set debug off
+show voicemail users=voicemail show users
+show voicemail zones=voicemail show zones
+iax2 trunk debug=iax2 set debug trunk
+iax2 jb debug=iax2 set debug jb
+iax2 no debug=iax2 set debug off
+iax2 no trunk debug=iax2 set debug trunk off
+iax2 no jb debug=iax2 set debug jb off
+show agents=agent show
+show agents online=agent show online
+show memory allocations=memory show allocations
+show memory summary=memory show summary
+show version=core show version
+show version files=core show file version
+show profile=core show profile
+clear profile=core clear profile
+soft hangup=channel request hangup
+
+[asterisk_1dot2](asterisk_1dot2_tpl)
+; add any additional custom commands you want below here, for example:
+;die quickly=stop now
+
+[asterisk_1dot4_tpl](!)
+cdr status=cdr show status
+rtp debug=rtp set debug on
+rtcp debug=rtcp set debug on
+rtcp stats=rtcp set stats on
+stun debug=stun set debug on
+udptl debug=udptl set debug on
+core show globals=dialplan show globals
+core set global=dialplan set global
+core set chanvar=dialplan set chanvar
+agi dumphtml=agi dump html
+ael debug=ael set debug
+funcdevstate list=devstate list
+sip history=sip set history on
+skinny debug=skinny set debug on
+mgcp set debug=mgcp set debug on
+abort shutdown=core abort shutdown
+stop now=core stop now
+stop gracefully=core stop gracefully
+stop when convenient=core stop when convenient
+restart now=core restart now
+restart gracefully=core restart gracefully
+restart when convenient=core restart when convenient
+soft hangup=channel request hangup
+
+[asterisk_1dot4](asterisk_1dot4_tpl)
+; add any additional custom commands you want below here.
+
+[asterisk_11_tpl](!)
+jabber list nodes=xmpp list nodes
+jabber purge nodes=xmpp purge nodes
+jabber delete node=xmpp delete node
+jabber create collection=xmpp create collection
+jabber create leaf=xmpp create leaf
+jabber set debug=xmpp set debug
+jabber show connections=xmpp show connections
+jabber show buddies=xmpp show buddies
+features reload=module reload features
+
+[asterisk_11](asterisk_11_tpl)
+; add any additional custom commands you want below here.

version_1.8.2/pabx/etc/asterisk/cli_permissions.conf

@@ -0,0 +1,82 @@
+;
+; CLI permissions configuration example for Asterisk
+;
+; All the users that you want to connect with asterisk using
+; rasterisk, should have write/read access to the
+; asterisk socket (asterisk.ctl). You could change the permissions
+; of this file in 'asterisk.conf' config parameter: 'astctlpermissions' (0666)
+; found on the [files] section.
+;
+; general options:
+;
+; default_perm = permit | deny
+;                This is the default permissions to apply for a user that
+;                does not has a permissions definided.
+;
+; user options:
+; permit = <command name> | all		; allow the user to run 'command' |
+;					; allow the user to run 'all' the commands
+; deny = <command name> | all		; disallow the user to run 'command' |
+;					; disallow the user to run 'all' commands.
+;
+
+[general]
+
+default_perm=permit	; To leave asterisk working as normal
+			; we should set this parameter to 'permit'
+;
+; Follows the per-users permissions configs.
+;
+; This list is read in the sequence that is being written, so
+; In this example the user 'eliel' is allow to run only the following
+; commands:
+;          sip show peer
+;          core set debug
+;          core set verbose
+; If the user is not specified, the default_perm option will be apply to
+; every command.
+;
+; Notice that you can also use regular expressions to allow or deny access to a
+; certain command like: 'core show application D*'. In this example the user will be
+; allowed to view the documentation for all the applications starting with 'D'.
+; Another regular expression could be: 'channel originate SIP/[0-9]* extension *'
+; allowing the user to use 'channel originate' on a sip channel and with the 'extension'
+; parameter and avoiding the use of the 'application' parameter.
+;
+; We can also use the templates syntax:
+; [supportTemplate](!)
+; deny=all
+; permit=sip show       ; all commands starting with 'sip show' will be allowed
+; permit=core show
+;
+; You can specify permissions for a local group instead of a user,
+; just put a '@' and we will know that is a group.
+; IMPORTANT NOTE: Users permissions overwrite group permissions.
+;
+;[@adm]
+;deny=all
+;permit=sip
+;permit=core
+;
+;
+;[eliel]
+;deny=all
+;permit=sip show peer
+;deny=sip show peers
+;permit=core set
+;
+;
+;User 'tommy' inherits from template 'supportTemplate':
+;	deny=all
+;	permit=sip show
+;	permit=core show
+;[tommy](supportTemplate)
+;permit=core set debug
+;permit=dialplan show
+;
+;
+;[mark]
+;deny=all
+;permit=all
+;
+;

version_1.8.2/pabx/etc/asterisk/codecs.conf

@@ -0,0 +1,206 @@
+[speex]
+; CBR encoding quality [0..10]
+; used only when vbr = false
+quality => 3
+
+; codec complexity [0..10]
+; tradeoff between cpu/quality
+complexity => 2
+
+; perceptual enhancement [true / false]
+; improves clarity of decoded speech
+enhancement => true
+
+; voice activity detection [true / false]
+; reduces bitrate when no voice detected, used only for CBR
+; (implicit in VBR/ABR)
+vad => true
+
+; variable bit rate [true / false]
+; uses bit rate proportionate to voice complexity
+vbr => true
+
+; available bit rate [bps, 0 = off]
+; encoding quality modulated to match this target bit rate
+; not recommended with dtx or pp_vad - may cause bandwidth spikes
+abr => 0
+
+; VBR encoding quality [0-10]
+; floating-point values allowed
+vbr_quality => 4
+
+; discontinuous transmission [true / false]
+; stops transmitting completely when silence is detected
+; pp_vad is far more effective but more CPU intensive
+dtx => false
+
+; preprocessor configuration
+; these options only affect Speex v1.1.8 or newer
+
+; enable preprocessor [true / false]
+; allows dsp functionality below but incurs CPU overhead
+preprocess => false
+
+; preproc voice activity detection [true / false]
+; more advanced equivalent of DTX, based on voice frequencies
+pp_vad => false
+
+; preproc automatic gain control [true / false]
+pp_agc => false
+pp_agc_level => 8000
+
+; preproc denoiser [true / false]
+pp_denoise => false
+
+; preproc dereverb [true / false]
+pp_dereverb => false
+pp_dereverb_decay => 0.4
+pp_dereverb_level => 0.3
+
+
+[plc]
+; for all codecs which do not support native PLC
+; this determines whether to perform generic PLC
+; there is a minor performance penalty for this
+genericplc => true
+
+; Generate custom formats for formats requiring attributes.
+; After defining the custom format, the name used in defining
+; the format can be used throughout Asterisk in the format 'allow'
+; and 'disallow' options.
+;
+; Example: silk8 is a predefined custom format in this config file.
+; Once this config file is loaded, silk8 can be used anywhere a
+; peer's codec capabilities are defined.
+;
+; In sip.conf 'silk8' can be defined as a capability for a peer.
+; [peer1]
+; type=peer
+; host=dynamic
+; disallow=all
+; allow=silk8 ;custom codec defined in codecs.conf
+;
+; LIMITATIONS
+; Custom formats can only be defined at startup.  Any changes to this
+; file made after startup will not take into effect until after Asterisk
+; is restarted.
+;
+
+; Default Custom SILK format definitions, only one custom SILK format per
+; sample rate is allowed.
+[silk8]
+type=silk
+samprate=8000
+fec=true           ; turn on or off encoding with forward error correction.
+                   ; On recommended, off by default.
+packetloss_percentage=10   ; Estimated packet loss percentage in uplink direction.  This
+                           ; affects how much redundancy is built in when using fec.
+                           ; The higher the percentage, the larger amount of bandwidth is
+                           ; used. Default is 0%, 10% is recommended when fec is in use.
+
+maxbitrate=10000    ; Use the table below to make sure a useful bitrate is choosen
+                    ; for maxbitrate.  If not set or value is not within the bounds
+                    ; of the encoder, a default value is chosen.
+                    ;
+                    ;  sample rate | bitrate range
+                    ;  8khz        | 5000 - 20000 bps
+                    ;  12khz       | 7000 - 25000 bps
+                    ;  16khz       | 8000 - 30000 bps
+                    ;  24khz       | 20000- 40000 bps
+                    ;
+;dtx=true       ; Encode using discontinuous transmission mode or not. Turning this
+                ; on will save bandwidth during periods of silence at the cost of
+                ; increased computational complexity.  Off by default.
+
+[silk12]
+type=silk
+samprate=12000
+maxbitrate=12000
+fec=true
+packetloss_percentage=10;
+
+[silk16]
+type=silk
+samprate=16000
+maxbitrate=20000
+fec=true
+packetloss_percentage=10;
+
+[silk24]
+type=silk
+samprate=24000
+maxbitrate=30000
+fec=true
+packetloss_percentage=10;
+
+
+; Default custom CELT codec definitions. Only one custom CELT definition is allowed
+; per a sample rate.
+;[celt44]
+;type=celt
+;samprate=44100  ; The samplerate in hz. This option is required.
+;framesize=480   ; The framesize option represents the duration of each frame in samples.
+                 ; This must be a factor of 2.  This option is only advertised in an SDP
+                 ; when it is set.  Otherwise a default of framesize of 480 is assumed
+                 ; internally
+
+;[celt48]
+;type=celt
+;samprate=48000
+
+;[celt32]
+;type=celt
+;samprate=32000
+
+;============================ OPUS Section Options ============================
+;
+;[opus]
+;type= ; Must be of type "opus" (default: "")
+;packet_loss= ; Encoder's packet loss percentage. Can be any number between 0
+              ; and 100, inclusive. A higher value results in more loss
+              ; resistance. (default: 0)
+;complexity= ; Encoder's computational complexity. Can be any number between 0
+             ; and 10, inclusive. Note, 10 equals the highest complexity.
+             ; (default: 10)
+;max_bandwidth= ; Encoder's maximum bandwidth allowed. Sets an upper bandwidth
+                ; bound on the encoder. Can be any of the following: narrow,
+                ; medium, wide, super_wide, full. (default: full)
+;signal= ; Encoder's signal type. Aids in mode selection on the encoder: Can
+         ; be any of the following: auto, voice, music. (default: auto)
+;application= ; Encoder's application type. Can be any of the following: voip,
+              ; audio, low_delay. (default: voip)
+;max_playback_rate= ; Override the maximum playback rate in the offer's SDP.
+                    ; Any value between 8000 and 48000 (inclusive) is valid,
+                    ; however typically it should match one of the usual opus
+                    ; bandwidths. A value of "sdp" is also allowed. When set
+                    ; to "sdp" then the value from the offer's SDP is used.
+                    ; (default: "sdp")
+;bitrate= ; Override the maximum average bitrate in the offer's SDP. Any value
+          ; between 500 and 512000 is valid. The following values are also
+          ; allowed: auto, max, sdp. When set to "sdp" then the value from
+          ; the offer's sdp is used. (default: "sdp")
+;cbr= ; Override the constant bit rate parameter in the offer's SDP. A value of
+      ; 0/false/no represents a variable bit rate whereas 1/true/yes represents
+      ; a constant bit rate. A value of "sdp" is also allowed. When set to "sdp"
+      ; then the value from the offer's sdp is used. (default: "sdp")
+;fec= ; Override the use inband fec parameter in the offer's SDP. A value of
+      ; 0/false/no represents disabled whereas 1/true/yes represents enabled.
+      ; A value of "sdp" is also allowed. When set to "sdp" then the value from
+      ; the offer's sdp is used. (default: "sdp")
+;dtx= ; Override the use dtx parameter in the offer's SDP. A value of 0/false/no
+      ; represents disabled whereas 1/true/yes represents enabled. A value of
+      ; "sdp" is also allowed. When set to "sdp" then the value from the offer's
+      ; sdp is used. (default: "sdp")
+
+;=============================== OPUS Examples ================================
+;
+;[opus]
+;type=opus
+;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder
+;fec=no ; Force no inband fec on the encoder (i.e don't use what's on the SDP)
+
+;[myopus]
+;type=opus
+;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz
+;                   ; audio bandwidth at 16Khz sample rate)
+;cbr=yes ; Force a constant bit rate (i.e don't use what's on the SDP)

version_1.8.2/pabx/etc/asterisk/confbridge.conf

@@ -0,0 +1,12 @@
+[general]
+[default_user_simplesip]
+type=user
+dtmf_passthrough=no
+dsp_silence_threshold=2500
+dsp_drop_silence=yes
+
+[simplesip_menu]
+type=menu
+1=dialplan_exec(adiciona-conferencia,1,1)
+2=admin_kick_last
+9=admin_toggle_mute_participants

version_1.8.2/pabx/etc/asterisk/config_test.conf

@@ -0,0 +1,46 @@
+; Config to test config parsing
+; global and item have values that differ from defaults
+; global_defaults and item_defualts are to show all defaults are set
+; there should be an option for every default type, and a custom type
+
+[global]
+intopt=-1
+uintopt=1
+timelenopt1=1ms
+timelenopt2=1s
+timelenopt3=1m
+timelenopt4=1h
+doubleopt=0.1
+sockaddropt=1.2.3.4:1234
+boolopt=true
+boolflag1=true
+boolflag2=false
+boolflag3=true
+deny=0.0.0.0/0
+permit=1.2.3.4/32
+codecopt=!all,ulaw,g729
+stropt=test
+customopt=yes
+
+[global_defaults]
+
+[item]
+intopt=-1
+uintopt=1
+timelenopt1=1
+timelenopt2=1
+timelenopt3=1
+timelenopt4=1
+doubleopt=0.1
+sockaddropt=1.2.3.4:1234
+boolopt=true
+boolflag1=true
+boolflag2=false
+boolflag3=true
+acldenyopt=0.0.0.0/0
+aclpermitopt=1.2.3.4/32
+codecopt=!all,ulaw,g729
+stropt=test
+customopt=yes
+
+[item_defaults]

version_1.8.2/pabx/etc/asterisk/console.conf

@@ -0,0 +1,97 @@
+;
+; Configuration for chan_console, a cross-platform console channel driver.
+;
+
+[general]
+
+; Set this option to "yes" to enable automatically answering calls on the
+; console.  This is very useful if the console is used as an intercom.
+; The default value is "no".
+;
+;autoanswer = no
+
+; Set the default context to use for outgoing calls.  This can be overridden by
+; dialing some extension@context, unless the overridecontext option is enabled.
+; The default is "default".
+;
+;context = default
+
+; Set the default extension to use for outgoing calls.  The default is "s".
+;
+;extension = s
+
+; Set the default CallerID for created channels.
+;
+;callerid = MyName Here <(256) 428-6000>
+
+; Set the default language for created channels.
+;
+;language = en
+
+; If you set overridecontext to 'yes', then the whole dial string
+; will be interpreted as an extension, which is extremely useful
+; to dial SIP, IAX and other extensions which use the '@' character.
+; The default is "no".
+;
+;overridecontext = no    ; if 'no', the last @ will start the context
+                        ; if 'yes' the whole string is an extension.
+
+
+; Default Music on Hold class to use when this channel is placed on hold in
+; the case that the music class is not set on the channel with
+; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
+; putting this one on hold did not suggest a class to use.
+;
+;mohinterpret=default
+
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
+                              ; Console channel. Defaults to "no". An enabled jitterbuffer will
+                              ; be used only if the sending side can create and the receiving
+                              ; side can not accept jitter. The Console channel can't accept jitter,
+                              ; thus an enabled jitterbuffer on the receive Console side will always
+                              ; be used if the sending side can create jitter.
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a Console
+                              ; channel. Two implementations are currently available - "fixed"
+                              ; (with size always equals to jbmax-size) and "adaptive" (with
+                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+; ----------------------------------------------------------------------------------
+
+
+;
+; Any configuration context defined beyond the [general] section configures
+; specific devices for use.
+;
+
+[default]
+input_device = default       ; When configuring an input device and output device,
+output_device = default      ; use the name that you see when you run the "console
+                             ; list available" CLI command.  If you say "default", the
+                             ; system default input and output devices will be used.
+autoanswer = no
+context = default
+extension = s
+callerid = MyName Here <(256) 428-6000>
+language = en
+overridecontext = no
+mohinterpret = default
+active = yes                 ; This option should only be set for one console.
+                             ; It means that it is the active console to be
+                             ; used from the Asterisk CLI.

version_1.8.2/pabx/etc/asterisk/dahdi_guiread.conf

@@ -0,0 +1,5 @@
+
+[general]
+
+#include "../zaptel.conf"
+

version_1.8.2/pabx/etc/asterisk/dahdi_scan.conf

@@ -0,0 +1,11 @@
+[1]
+active=yes
+alarms=UNCONFIGURED
+description=ZTDUMMY/1 (source: HRtimer) 1
+name=ZTDUMMY/1
+manufacturer=
+devicetype=Zaptel Dummy Timing Driver
+location=
+basechan=1
+totchans=0
+irq=0

version_1.8.2/pabx/etc/asterisk/dbsep.conf

@@ -0,0 +1,34 @@
+#
+# Configuration file for dbsep.cgi
+#
+# The purpose of this file is to provide realtime access to a database,
+# possibly through ODBC, without needing to load the ODBC drivers into
+# Asterisk, since there are several backend drivers which are rather
+# buggy.
+#
+# We accomplish this separation by using the res_config_curl realtime
+# driver to connect to a server running dbsep.cgi (or another, which
+# implements the same protocol).
+#
+# This file contains the information necessary to configure dbsep.cgi.
+#
+#
+# Once installed to a web server, you'll need to preload func_curl.so
+# and res_config_curl.so in modules.conf and configure extconfig.conf:
+#
+# voicemail => curl,http://server/path/to/dbsep.cgi/voicemail
+# sippeers => curl,http://server/path/to/dbsep.cgi/sippeers
+#
+
+# The Data Source Name, as specified by the Perl DBI module.
+# Typically, this will be along the lines of 'DBI:mysql:astdbname[:dbhostname]' or 'DBI:Pg:dbname=astdbname;hostname=dbhostname'
+dsn=somedsn
+
+# Connected database user
+dbuser=someuser
+
+# And its password
+dbpass=password
+
+# For most databases, this is fine.  Set to 'no' for Sybase or MS SQL Server.
+backslash_is_escape=yes

version_1.8.2/pabx/etc/asterisk/digivoice.conf

@@ -0,0 +1,27 @@
+[general]
+allow_slinear=0
+allow_ulaw=1
+allow_alaw=1
+allow_gsm=0
+consolelanguage=br
+
+[allportsconfig]
+default_callprogress=cp_default.cfg
+detectiontype=1
+dialtype=1
+afterdialpause=1000
+ringbacktone1=tone1
+ringbacktone2=tone1
+ringbacktimes=1000,4000,1000,4000
+busytone1=tone1
+busytone2=tone1
+busytimes=250,250,250,250
+subchannelringtone1=tone1
+subchannelringtone2=tone1
+subchannelringtimes=100,100,100,5000
+
+[groups]
+
+[port_config]
+
+[e1_config]

version_1.8.2/pabx/etc/asterisk/dnsmgr.conf

@@ -0,0 +1,5 @@
+[general]
+enable=no		; enable creation of managed DNS lookups
+			;   default is 'no'
+;refreshinterval=1200	; refresh managed DNS lookups every <n> seconds
+			;   default is 300 (5 minutes)

version_1.8.2/pabx/etc/asterisk/dsp.conf

@@ -0,0 +1,42 @@
+[default]
+;
+; Length of sound (in milliseconds) before a period of silence is considered
+; to be a change from talking to silence or a period of noise converts silence
+; to talking.  [default=256]
+;
+;silencethreshold=256
+
+; DTMF Reverse Twist and Normal Twist is the difference in power between the row and column energies.
+;
+; Normal Twist is where the row energy is greater than the column energy.
+; Reverse Twist is where the column energy is greater.
+;
+; Power level difference between frequencies for different Administrations/RPOAs
+;               Power Gain              equiv
+;               normal  reverse         dB's
+; AT&T(default) 6.31    2.51            8dB(normal), 4dB(reverse)
+; NTT           3.16    3.16            Max. 5dB
+; Danish        3.98    3.98            Max. 6dB
+; Australian    10.0    10.0            Max. 10dB
+; Brazilian     7.94    7.94            Max. 9dB
+; ETSI          3.98    3.98            Max. 6dB
+
+;previous version compatible AT&T values
+; RADIO_RELAX disabled, and relaxdtmf=no
+;               6.30    2.50            7.99dB(normal), 3.98dB(reverse)
+; RADIO_RELAX disabled, and relaxdtmf=yes
+;               6.30    4.00            7.99dB(normal), 6.02dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=no
+;               6.30    2.50            7.99dB(normal), 3.984dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=yes
+;               6.30    6.50            7.99dB(normal), 8.13dB(reverse)
+
+;If you don't know what these mean, don't change them.
+;dtmf_normal_twist=6.31
+;dtmf_reverse_twist=2.51
+;relax_dtmf_normal_twist=6.31
+;relax_dtmf_reverse_twist=3.98
+
+;successive number hits/misses of 12.75ms before a digit/nodigit is considered valid
+;dtmf_hits_to_begin=2
+;dtmf_misses_to_end=3

version_1.8.2/pabx/etc/asterisk/dundi.conf

@@ -0,0 +1,268 @@
+;
+; DUNDi configuration file
+;
+; For more information about DUNDi, see http://www.dundi.com
+;
+;
+[general]
+;
+; The "general" section contains general parameters relating
+; to the operation of the dundi client and server.
+;
+; The first part should be your complete contact information
+; should someone else in your peer group need to contact you.
+;
+;department=Your Department
+;organization=Your Company, Inc.
+;locality=Your City
+;stateprov=ST
+;country=US
+;email=your@email.com
+;phone=+12565551212
+;
+;
+; Specify bind address and port number.  Default is
+; 4520
+;
+;bindaddr=0.0.0.0
+;port=4520
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
+;tos=ef
+;
+; Our entity identifier (Should generally be the MAC address of the
+; machine it's running on.  Defaults to the first eth address, but you
+; can override it here, as long as you set it to the MAC of *something*
+; you own!)  The EID can be overridden by a setting in asterisk.conf,
+; or by setting this option.
+;
+;entityid=00:07:E9:3B:76:60
+;
+; Peers shall cache our query responses for the specified time,
+; given in seconds. Default is 3600.
+;
+;cachetime=3600
+;
+; This defines the max depth in which to search the DUNDi system.
+; Note that the maximum time that we will wait for a response is
+; (2000 + 200 * ttl) ms.
+;
+ttl=32
+;
+; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set
+; to yes, then we cancel the whole thing (that's enough time for one
+; retransmission only).  This is used to keep things from stalling for a long
+; time for a host that is not available, but would be ill advised for bad
+; connections.  In addition to 'yes' or 'no' you can also specify a number
+; of milliseconds.  See 'qualify' for individual peers to turn on for just
+; a specific peer.
+;
+autokill=yes
+;
+; pbx_dundi creates a rotating key called "secret", under the family
+; 'secretpath'.  The default family is dundi (resulting in
+; the key being held at dundi/secret).
+;
+;secretpath=dundi
+;
+; The 'storehistory' option (also changeable at runtime with
+; 'dundi store history' and 'dundi no store history') will
+; cause the DUNDi engine to keep track of the last several
+; queries and the amount of time each query took to execute
+; for the purpose of tracking slow nodes.  This option is
+; off by default due to performance impacts.
+;
+;storehistory=yes
+
+[mappings]
+;
+; The "mappings" section maps DUNDi contexts
+; to contexts on the local asterisk system.  Remember
+; that numbers that are made available under the e164
+; DUNDi context are regulated by the DUNDi General Peering
+; Agreement (GPA) if you are a member of the DUNDi E.164
+; Peering System.
+;
+; dundi_context => local_context,weight,tech,dest[,options]]
+;
+; 'dundi_context' is the name of the context being requested
+; within the DUNDi request
+;
+; 'local_context' is the name of the context on the local system
+; in which numbers can be looked up for which responses shall be given.
+;
+; 'weight' is the weight to use for the responses provided from this
+; mapping.  The number must be >= 0 and < 60000.  Since it is totally
+; valid to receive multiple responses to a query, responses received
+; with a lower weight are tried first.  Note that the weight has a
+; special meaning in the e164 context - see the GPA for more details.
+;
+; 'tech' is the technology to use (IAX, SIP, H323)
+;
+; 'dest' is the destination to supply for reaching that number.  The
+; following variables can be used in the destination string and will
+; be automatically substituted:
+; ${NUMBER}: The number being requested
+; ${IPADDR}: The IP address to connect to
+; ${SECRET}: The current rotating secret key to be used
+;
+; Further options may include:
+;
+; nounsolicited:  No unsolicited calls of any type permitted via this
+;                 route
+; nocomunsolicit: No commercial unsolicited calls permitted via
+;                 this route
+; residential:    This number is known to be a residence
+; commercial:     This number is known to be a business
+; mobile:         This number is known to be a mobile phone
+; nocomunsolicit: No commercial unsolicited calls permitted via
+;                 this route
+; nopartial:      Do not search for partial matches
+;
+; There *must* exist an entry in mappings for DUNDi to respond
+; to any request, although it may be empty.
+;
+;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+
+;digexten => default,0,IAX2,guest@lappy/${NUMBER}
+;asdf =>
+
+;
+; Weights for mappings can be set a few different ways:
+;
+; 1) It can be set as a static number.
+;testmap1 => context1,222,IAX2,guest@peer1/${NUMBER}
+;
+; 2) It can be an Asterisk global variable.
+;testmap2 => context2,${DUNDITESTVAR},IAX2,guest@peer2${NUMBER}
+;
+; 3) It can be retrieved using a dialplan function.  This can be extremely
+;    useful if you want to let an external script decide what the weight
+;    in a response shouuld be.
+;testmap3 => context3,${SHELL(echo 123)},IAX2,guest@peer3/${NUMBER}
+;
+; The built in variables ${SECRET}, ${IPADDR} and ${NUMBER} can also be
+; passed to the weight. For example, you could pass the ${NUMBER} value
+; to your SHELL() script and use that to dynamically return a weight.
+;
+; Note than when using a global variable or dialplan function to set the
+; weight for a mapping, that response caching should be disabled if you
+; plan for these values to change frequently at all.  If the results are
+; cached, then any change in value will not take effect until the cache
+; has expired.
+;
+
+;
+; The remaining sections represent the peers
+; that we fundamentally trust.  The section name
+; represents the name and optionally at a specific
+; DUNDi context if you want the trust to be established
+; for only a specific DUNDi context.
+;
+; inkey - What key they will be authenticating to us with
+;
+; outkey - What key we use to authenticate to them
+;
+; host - What their host is
+;
+; port - The port where their host is listening (default: 4520)
+;
+; order - What search order to use.  May be 'primary', 'secondary',
+;         'tertiary' or 'quartiary'.  In large systems, it is beneficial
+;         to only query one up-stream host in order to maximize caching
+;         value.  Adding one with primary and one with secondary gives you
+;         redundancy without sacrificing performance.
+;
+; include - Includes this peer when searching a particular context
+;           for lookup (set "all" to perform all lookups with that
+;           host.  This is also the context in which peers are permitted
+;           to precache.
+;
+; noinclude - Disincludes this peer when searching a particular context
+;             for lookup (set "all" to perform no lookups with that
+;             host.
+;
+; permit - Permits this peer to search a given DUNDi context on
+;          the local system.  Set "all" to permit this host to
+;          lookup all contexts.  This is also a context for which
+;          we will create/forward PRECACHE commands.
+;
+; deny -   Denies this peer to search a given DUNDi context on
+;          the local system.  Set "all" to deny this host to
+;          lookup all contexts.
+;
+; model - inbound, outbound, or symmetric for whether we receive
+;         requests only, transmit requests only, or do both.
+;
+; precache - Utilize/Permit precaching with this peer (to pre
+;            cache means to provide an answer when no request
+;            was made and is used so that machines with few
+;            routes can push those routes up a to a higher level).
+;            outgoing means we send precache routes to this peer,
+;            incoming means we permit this peer to send us
+;            precache routes.  symmetric means we do both.
+;
+; Note: You cannot mix symmetric/outbound model with symmetric/inbound
+; precache, nor can you mix symmetric/inbound model with symmetric/outbound
+; precache.
+;
+;
+; The '*' peer is special and matches an unspecified entity
+;
+
+;
+; Sample Primary e164 DUNDi peer
+;
+;[00:50:8B:F3:75:BB]
+;model = symmetric
+;host = 64.215.96.114
+;inkey = digium
+;outkey = misery
+;include = e164
+;permit = e164
+;qualify = yes
+
+;
+; Sample Secondary e164 DUNDi peer
+;
+;[00:A0:C9:96:92:84]
+;model = symmetric
+;host = misery.digium.com
+;inkey = misery
+;outkey = ourkey
+;include = e164
+;permit = e164
+;qualify = yes
+;order = secondary
+
+;
+; Sample "push mode" downstream host
+;
+;[00:0C:76:96:75:28]
+;model = inbound
+;host = dynamic
+;precache = inbound
+;inkey = littleguy
+;outkey = ourkey
+;include = e164	; In this case used only for precaching
+;permit = e164
+;qualify = yes
+
+;
+; Sample "push mode" upstream host
+;
+;[00:07:E9:3B:76:60]
+;model = outbound
+;precache = outbound
+;host = 216.207.245.34
+;register = yes
+;inkey = dhcp34
+;permit = all ; In this case used only for precaching
+;include = all
+;qualify = yes
+;outkey=foo
+
+;[*]
+;

version_1.8.2/pabx/etc/asterisk/enum.conf

@@ -0,0 +1,22 @@
+;
+; ENUM Configuration for resolving phone numbers over DNS
+;
+; Sample config for Asterisk
+; This file is reloaded at "module reload enum" in the CLI
+;
+[general]
+;
+; The search list for domains may be customized.  Domains are searched
+; in the order they are listed here.
+;
+search => e164.arpa
+;
+; If you'd like to use the E.164.org public ENUM registry in addition
+; to the official e164.arpa one, uncomment the following line
+;
+;search => e164.org
+;
+; As there are more H323 drivers available you have to select to which
+; drive a H323 URI will map. Default is "H323".
+;
+h323driver => H323

version_1.8.2/pabx/etc/asterisk/extconfig.conf

@@ -0,0 +1,111 @@
+;
+; Static and realtime external configuration
+; engine configuration
+;
+; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
+; for basic table formatting information.
+;
+[settings]
+;
+; Static configuration files:
+;
+; file.conf => driver,database[,table[,priority]]
+;
+; maps a particular configuration file to the given
+; database driver, database and table (or uses the
+; name of the file as the table if not specified)
+;
+; Uncomment to load queues.conf via the odbc engine.
+;
+;queues.conf => odbc,asterisk,ast_config
+;extensions.conf => sqlite,asterisk,ast_config
+;
+; The following files CANNOT be loaded from Realtime storage:
+;	asterisk.conf
+;	extconfig.conf (this file)
+;	logger.conf
+;
+; Additionally, the following files cannot be loaded from
+; Realtime storage unless the storage driver is loaded
+; early using 'preload' statements in modules.conf:
+;	manager.conf
+;	cdr.conf
+;	rtp.conf
+;
+; Named ACLs specified in realtime also can not be used
+; from manager.conf unless the storage driver is preloaded.
+; Attempting to use a realtime stored named ACL before the
+; driver is loaded will result in an invalid ACL which
+; rejects all addresses.
+;
+; Realtime configuration engine
+;
+; maps a particular family of realtime
+; configuration to a given database driver,
+; database and table (or uses the name of
+; the family if the table is not specified
+;
+;example => odbc,asterisk,alttable,1
+;example => mysql,asterisk,alttable,2
+;example2 => ldap,"dc=oxymium,dc=net",example2
+;
+; Additionally, priorities are now supported for use as failover methods
+; for retrieving realtime data.  If one connection fails to retrieve any
+; information, the next sequential priority will be tried next.  This
+; especially works well with ODBC connections, since res_odbc now caches
+; when connection failures occur and prevents immediately retrying those
+; connections until after a specified timeout.  Note:  priorities must
+; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
+; and 4, then 4 will be ignored, because there is no 3).
+;
+;
+; Possible driver backends:
+;
+; "odbc" is shown in the examples below, but is not the only valid realtime
+; engine.  Here are several of the possible options:
+;    odbc ... res_config_odbc
+;    sqlite ... res_config_sqlite
+;    sqlite3 ... res_config_sqlite3
+;    pgsql ... res_config_pgsql
+;    curl ... res_config_curl
+;    ldap ... res_config_ldap
+;    mysql ... res_config_mysql (available via add-ons in menuselect)
+;
+; Note: The res_config_pgsql and res_config_sqlite backends configure the
+; database used in their respective configuration files and ignore the
+; database name configured in this file.
+;
+;iaxusers => odbc,asterisk
+;iaxpeers => odbc,asterisk
+;sippeers => odbc,asterisk
+;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
+;ps_endpoints => odbc,asterisk
+;ps_auths => odbc,asterisk
+;ps_aors => odbc,asterisk
+;ps_domain_aliases => odbc,asterisk
+;ps_endpoint_id_ips => odbc,asterisk
+;ps_outbound_publishes => odbc,asterisk
+;ps_inbound_publications = odbc,asterisk
+;ps_asterisk_publications = odbc,asterisk
+;voicemail => odbc,asterisk
+;extensions => odbc,asterisk
+;meetme => mysql,general
+;queues => odbc,asterisk
+;queue_members => odbc,asterisk
+;queue_rules => odbc,asterisk
+;acls => odbc,asterisk
+;musiconhold => mysql,general
+;queue_log => mysql,general
+;
+;
+; While most dynamic realtime engines are automatically used when defined in
+; this file, 'extensions', distinctively, is not.  To activate dynamic realtime
+; extensions, you must turn them on in each respective context within
+; extensions.conf with a switch statement.  The syntax is:
+;      switch => Realtime/[[db_context@]tablename]/<opts>
+; The only option available currently is the 'p' option, which disallows
+; extension pattern queries to the database.  If you have no patterns defined
+; in a particular context, this will save quite a bit of CPU time.  However,
+; note that using dynamic realtime extensions is not recommended anymore as a
+; best practice; instead, you should consider writing a static dialplan with
+; proper data abstraction via a tool like func_odbc.

version_1.8.2/pabx/etc/asterisk/extensions.ael

@@ -0,0 +1,456 @@
+//
+// Example AEL config file
+//
+//
+// Static extension configuration file, used by
+// the pbx_ael module. This is where you configure all your
+// inbound and outbound calls in Asterisk.
+//
+// This configuration file is reloaded
+// - With the "ael reload" command in the CLI
+// - With the "reload" command (that reloads everything) in the CLI
+
+// The "Globals" category contains global variables that can be referenced
+// in the dialplan by using the GLOBAL dialplan function:
+//  ${GLOBAL(VARIABLE)}
+// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
+// Unix/Linux environmental variables are reached with the ENV dialplan
+// function: ${ENV(VARIABLE)}
+//
+
+// NOTE! NOTE! NOTE!
+// Asterisk by default will load both extensions.conf and extensions.ael files.
+// Upon loading these files the dialplans generated from both with be merged,
+// so you must make sure that you don't have any overlapping contexts or global
+// variables. If you do, then unexpected behavior may result when the data is
+// merged.
+// NOTE! NOTE! NOTE!
+
+globals {
+	CONSOLE-AEL="Console/dsp"; 		// Console interface for demo
+	//CONSOLE-AEL=Zap/1;
+	//CONSOLE-AEL=Phone/phone0;
+	IAXINFO-AEL=guest;				// IAXtel username/password
+	//IAXINFO-AEL="myuser:mypass";
+	OUTBOUND-TRUNK="Zap/g2";		// Trunk interface
+	//
+	// Note the 'g2' in the OUTBOUND-TRUNK variable above. It specifies which group (defined
+	// in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
+	// the specified group. The four possible options are:
+	//
+	// g: select the lowest-numbered non-busy DAHDI channel
+	//    (aka. ascending sequential hunt group).
+	// G: select the highest-numbered non-busy DAHDI channel
+	//    (aka. descending sequential hunt group).
+	// r: use a round-robin search, starting at the next highest channel than last
+	//    time (aka. ascending rotary hunt group).
+	// R: use a round-robin search, starting at the next lowest channel than last
+	//    time (aka. descending rotary hunt group).
+	//
+	OUTBOUND-TRUNKMSD=1;					// MSD digits to strip (usually 1 or 0)
+	//OUTBOUND-TRUNK2=IAX2/user:pass@provider;
+};
+
+//
+// Any category other than "General" and "Globals" represent
+// extension contexts, which are collections of extensions.
+//
+// Extension names may be numbers, letters, or combinations
+// thereof. If an extension name is prefixed by a '_'
+// character, it is interpreted as a pattern rather than a
+// literal.  In patterns, some characters have special meanings:
+//
+//   X - any digit from 0-9
+//   Z - any digit from 1-9
+//   N - any digit from 2-9
+//   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+//   . - wildcard, matches anything remaining (e.g. _9011. matches
+//	anything starting with 9011 excluding 9011 itself)
+//   ! - wildcard, causes the matching process to complete as soon as
+//       it can unambiguously determine that no other matches are possible
+//
+// For example the extension _NXXXXXX would match normal 7 digit dialings,
+// while _1NXXNXXXXXX would represent an area code plus phone number
+// preceded by a one.
+//
+// Each step of an extension is ordered by priority, which must
+// always start with 1 to be considered a valid extension.  The priority
+// "next" or "n" means the previous priority plus one, regardless of whether
+// the previous priority was associated with the current extension or not.
+// The priority "same" or "s" means the same as the previously specified
+// priority, again regardless of whether the previous entry was for the
+// same extension.  Priorities may be immediately followed by a plus sign
+// and another integer to add that amount (most useful with 's' or 'n').
+// Priorities may then also have an alias, or label, in
+// parenthesis after their name which can be used in goto situations
+//
+// Contexts contain several lines, one for each step of each
+// extension, which can take one of two forms as listed below,
+// with the first form being preferred.  One may include another
+// context in the current one as well, optionally with a
+// date and time.  Included contexts are included in the order
+// they are listed.
+//
+//context name {
+//	exten-name => {
+//		application(arg1,arg2,...);
+//
+// 	Timing list for includes is
+//
+//   <time range>|<days of week>|<days of month>|<months>
+//
+//	includes {
+//		daytime|9:00-17:00|mon-fri|*|*;
+//      };
+//
+// 	ignorepat can be used to instruct drivers to not cancel dialtone upon
+// 	receipt of a particular pattern.  The most commonly used example is
+// 	of course '9' like this:
+//
+//	ignorepat => 9;
+//
+// 	so that dialtone remains even after dialing a 9.
+//};
+
+
+//
+// Sample entries for extensions.conf
+//
+//
+context ael-dundi-e164-canonical {
+	//
+	// List canonical entries here
+	//
+	// 12564286000 => &ael-std-exten(6000,IAX2/foo);
+	// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
+};
+
+context ael-dundi-e164-customers {
+	//
+	// If you are an ITSP or Reseller, list your customers here.
+	//
+	//_12564286000 => Dial(SIP/customer1);
+	//_12564286001 => Dial(IAX2/customer2);
+};
+
+context ael-dundi-e164-via-pstn {
+	//
+	// If you are freely delivering calls to the PSTN, list them here
+	//
+	//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
+	//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
+};
+
+context ael-dundi-e164-local {
+	//
+	// Context to put your dundi IAX2 or SIP user in for
+	// full access
+	//
+	includes {
+	 ael-dundi-e164-canonical;
+	 ael-dundi-e164-customers;
+	 ael-dundi-e164-via-pstn;
+	};
+};
+
+context ael-dundi-e164-switch {
+	//
+	// Just a wrapper for the switch
+	//
+
+	switches {
+		DUNDi/e164;
+	};
+};
+
+context ael-dundi-e164-lookup {
+	//
+	// Locally to lookup, try looking for a local E.164 solution
+	// then try DUNDi if we don't have one.
+	//
+	includes {
+		ael-dundi-e164-local;
+		ael-dundi-e164-switch;
+	};
+	//
+};
+
+//
+// DUNDi can also be implemented as a Macro instead of using
+// the Local channel driver.
+//
+macro ael-dundi-e164(exten) {
+//
+// ARG1 is the extension to Dial
+//
+	goto ${exten}|1;
+	return;
+};
+
+//
+// Here are the entries you need to participate in the IAXTEL
+// call routing system.  Most IAXTEL numbers begin with 1-700, but
+// there are exceptions.  For more information, and to sign
+// up, please go to www.gnophone.com or www.iaxtel.com
+//
+context ael-iaxtel700 {
+	_91700XXXXXXX => Dial(IAX2/${IAXINFO-AEL}@iaxtel.com/${EXTEN:1}@iaxtel);
+};
+
+//
+// The SWITCH statement permits a server to share the dialplan with
+// another server. Use with care: Reciprocal switch statements are not
+// allowed (e.g. both A -> B and B -> A), and the switched server needs
+// to be on-line or else dialing can be severly delayed.
+//
+context ael-iaxprovider {
+	switches {
+	// IAX2/user:[key]@myserver/mycontext;
+	};
+};
+
+context ael-trunkint {
+	//
+	// International long distance through trunk
+	//
+	includes {
+		ael-dundi-e164-lookup;
+	};
+	_9011. => {
+		&ael-dundi-e164(${EXTEN:4});
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunkld {
+	//
+	// Long distance context accessed through trunk
+	//
+	includes {
+		ael-dundi-e164-lookup;
+	};
+	_91NXXNXXXXXX => {
+		&ael-dundi-e164(${EXTEN:1});
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunklocal {
+	//
+	// Local seven-digit dialing accessed through trunk interface
+	//
+	_9NXXXXXX => {
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunktollfree {
+	//
+	// Long distance context accessed through trunk interface
+	//
+
+	_91800NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91888NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91877NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91866NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+};
+
+context ael-international {
+	//
+	// Master context for international long distance
+	//
+	ignorepat => 9;
+	includes {
+		ael-longdistance;
+		ael-trunkint;
+	};
+};
+
+context ael-longdistance {
+	//
+	// Master context for long distance
+	//
+	ignorepat => 9;
+	includes {
+		ael-local;
+		ael-trunkld;
+	};
+};
+
+context ael-local {
+	//
+	// Master context for local, toll-free, and iaxtel calls only
+	//
+	ignorepat => 9;
+	includes {
+		ael-default;
+		ael-trunklocal;
+		ael-iaxtel700;
+		ael-trunktollfree;
+		ael-iaxprovider;
+	};
+};
+
+//
+// You can use an alternative switch type as well, to resolve
+// extensions that are not known here, for example with remote
+// IAX switching you transparently get access to the remote
+// Asterisk PBX
+//
+// switch => IAX2/user:password@bigserver/local
+//
+// An "lswitch" is like a switch but is literal, in that
+// variable substitution is not performed at load time
+// but is passed to the switch directly (presumably to
+// be substituted in the switch routine itself)
+//
+// lswitch => Loopback/12${EXTEN}@othercontext
+//
+// An "eswitch" is like a switch but the evaluation of
+// variable substitution is performed at runtime before
+// being passed to the switch routine.
+//
+// eswitch => IAX2/context@${CURSERVER}
+
+
+macro ael-std-exten-ael( ext , dev ) {
+        Dial(${dev}/${ext},20);
+        switch(${DIALSTATUS}) {
+        case BUSY:
+                Voicemail(${ext},b);
+                break;
+        default:
+                Voicemail(${ext},u);
+        };
+        catch a {
+                VoiceMailMain(${ext});
+                return;
+        };
+	return;
+};
+
+context ael-demo {
+	s => {
+		Wait(1);
+		Answer();
+		Set(TIMEOUT(digit)=5);
+		Set(TIMEOUT(response)=10);
+restart:
+		Background(demo-congrats);
+instructions:
+		for (x=0; ${x} < 3; x=${x} + 1) {
+			Background(demo-instruct);
+			WaitExten();
+		};
+	};
+	2 => {
+		Background(demo-moreinfo);
+		goto s|instructions;
+	};
+	3 => {
+		Set(LANGUAGE()=fr);
+		goto s|restart;
+	};
+	1000 => {
+		goto ael-default|s|1;
+	};
+	500 => {
+		Playback(demo-abouttotry);
+		Dial(IAX2/guest@misery.digium.com/s@default);
+		Playback(demo-nogo);
+		goto s|instructions;
+	};
+	600 => {
+		Playback(demo-echotest);
+		Echo();
+		Playback(demo-echodone);
+		goto s|instructions;
+	};
+	_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
+	8500 => {
+		VoicemailMain();
+		goto s|instructions;
+	};
+	# => {
+		Playback(demo-thanks);
+		Hangup();
+	};
+	t => goto #|1;
+	i => Playback(invalid);
+};
+
+
+//
+// If you wish to use AEL for your default context, remove it
+// from extensions.conf (or change its name or comment it out)
+// and then uncomment the one here.
+//
+
+context ael-default {
+
+// By default we include the demo.  In a production system, you
+// probably don't want to have the demo there.
+
+	includes {
+		ael-demo;
+	};
+//
+// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
+// Note that you must have a [sipprovider] section in sip.conf whereas
+// the otherprovider.net example does not require such a peer definition
+//
+//_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);
+//_42X. => Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
+
+// Real extensions would go here. Generally you want real extensions to be
+// 4 or 5 digits long (although there is no such requirement) and start with a
+// single digit that is fairly large (like 6 or 7) so that you have plenty of
+// room to overlap extensions and menu options without conflict.  You can alias
+// them with names, too, and use global variables
+
+// 6245  => {
+//		hint(SIP/Grandstream1&SIP/Xlite1,Joe Schmoe); // Channel hints for presence
+// 		Dial(SIP/Grandstream1,20,rt);                 // permit transfer
+//        Dial(${HINT}/5245},20,rtT);                    // Use hint as listed
+//        switch(${DIALSTATUS}) {
+//        case BUSY:
+//                Voicemail(6245,b);
+//				return;
+//        default:
+//                Voicemail(6245,u);
+//				return;
+//        };
+//       };
+
+// 6361 => Dial(IAX2/JaneDoe,,rm);                // ring without time limit
+// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
+// 6394 => Dial(Local/6275/n);                    // this will dial ${MARK}
+
+// 6275 => &ael-stdexten(6275,${MARK});           // assuming ${MARK} is something like DAHDI/2
+// mark => goto 6275|1;                          // alias mark to 6275
+// 6536 => &ael-stdexten(6236,${WIL});            // Ditto for wil
+// wil  => goto 6236|1;
+//
+// Some other handy things are an extension for checking voicemail via
+// voicemailmain
+//
+// 8500 => {
+//			VoicemailMain();
+//			Hangup();
+//	       };
+//
+// Or a conference room (you'll need to edit meetme.conf to enable this room)
+//
+// 8600 => Meetme(1234);
+//
+// Or playing an announcement to the called party, as soon it answers
+//
+// 8700 => Dial(${MARK},30,A(/path/to/my/announcemsg))
+//
+// For more information on applications, just type "show applications" at your
+// friendly Asterisk CLI prompt.
+//
+// 'show application <command>' will show details of how you
+// use that particular application in this file, the dial plan.
+//
+}

version_1.8.2/pabx/etc/asterisk/extensions.conf

@@ -0,0 +1,27 @@
+[general]
+#include extensions_general.conf
+#include extensions_ramais.conf
+#include extensions_disa.conf
+#include extensions_fila.conf
+#include extensions_fila_login.conf
+#include extensions_horarios.conf
+#include extensions_facilidades.conf
+#include extensions_grupos.conf
+#include extensions_ura.conf
+#include extensions_rotas_entrada.conf
+#include extensions_contextos.conf
+#include extensions_conferencia.conf
+#include extensions_anuncios.conf
+#include extensions_teste.conf
+#include extensions_rotas_interno.conf
+#include extensions_contextos_personalizados.conf
+#include extensions_callback.conf
+#include extensions_padrao.conf
+#include extensions_padrao_customizado.conf
+#include extensions_campanha.conf
+#include extensions_rotas_saida_campanha.conf
+#include extensions_rotas.conf
+#include extensions_transfere_campanha.conf
+#include extensions_pesquisa.conf
+#include extensions_contextos_custom.conf
+#include hints.conf

version_1.8.2/pabx/etc/asterisk/extensions.lua

@@ -0,0 +1,240 @@
+
+
+CONSOLE = "Console/dsp" -- Console interface for demo
+--CONSOLE = "DAHDI/1"
+--CONSOLE = "Phone/phone0"
+
+IAXINFO = "guest"       -- IAXtel username/password
+--IAXINFO = "myuser:mypass"
+
+TRUNK = "DAHDI/G2"
+TRUNKMSD = 1
+-- TRUNK = "IAX2/user:pass@provider"
+
+
+--
+-- Extensions are expected to be defined in a global table named 'extensions'.
+-- The 'extensions' table should have a group of tables in it, each
+-- representing a context.  Extensions are defined in each context.  See below
+-- for examples.
+--
+-- Extension names may be numbers, letters, or combinations thereof. If
+-- an extension name is prefixed by a '_' character, it is interpreted as
+-- a pattern rather than a literal.  In patterns, some characters have
+-- special meanings:
+--
+--   X - any digit from 0-9
+--   Z - any digit from 1-9
+--   N - any digit from 2-9
+--   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+--   . - wildcard, matches anything remaining (e.g. _9011. matches
+--       anything starting with 9011 excluding 9011 itself)
+--   ! - wildcard, causes the matching process to complete as soon as
+--       it can unambiguously determine that no other matches are possible
+--
+-- For example the extension _NXXXXXX would match normal 7 digit
+-- dialings, while _1NXXNXXXXXX would represent an area code plus phone
+-- number preceded by a one.
+--
+-- If your extension has special characters in it such as '.' and '!' you must
+-- explicitly make it a string in the tabale definition:
+--
+--   ["_special."] = function;
+--   ["_special!"] = function;
+--
+-- There are no priorities.  All extensions to asterisk appear to have a single
+-- priority as if they consist of a single priority.
+--
+-- Each context is defined as a table in the extensions table.  The
+-- context names should be strings.
+--
+-- One context may be included in another context using the 'includes'
+-- extension.  This extension should be set to a table containing a list
+-- of context names.  Do not put references to tables in the includes
+-- table.
+--
+--   include = {"a", "b", "c"};
+--
+-- Channel variables can be accessed thorugh the global 'channel' table.
+--
+--   v = channel.var_name
+--   v = channel["var_name"]
+--   v.value
+--   v:get()
+--
+--   channel.var_name = "value"
+--   channel["var_name"] = "value"
+--   v:set("value")
+--
+--   channel.func_name(1,2,3):set("value")
+--   value = channel.func_name(1,2,3):get()
+--
+--   channel["func_name(1,2,3)"]:set("value")
+--   channel["func_name(1,2,3)"] = "value"
+--   value = channel["func_name(1,2,3)"]:get()
+--
+-- Note the use of the ':' operator to access the get() and set()
+-- methods.
+--
+-- Also notice the absence of the following constructs from the examples above:
+--   channel.func_name(1,2,3) = "value"  -- this will NOT work
+--   value = channel.func_name(1,2,3)    -- this will NOT work as expected
+--
+--
+-- Dialplan applications can be accessed through the global 'app' table.
+--
+--    app.Dial("DAHDI/1")
+--    app.dial("DAHDI/1")
+--    app["dial"]("DAHDI/1")
+--
+-- More examples can be found below.
+--
+-- An autoservice is automatically run while lua code is executing.  The
+-- autoservice can be stopped and restarted using the autoservice_stop() and
+-- autoservice_start() functions.  The autservice should be running before
+-- starting long running operations. The autoservice will automatically be
+-- stopped before executing applications and dialplan functions and will be
+-- restarted afterwards.  The autoservice_status() function can be used to
+-- check the current status of the autoservice and will return true if an
+-- autoservice is currently running.
+--
+-- Note about naming conflicts:
+-- Lua allows you to refer to table entries using the '.' notation,
+-- I.E. app.goto(something), only if the entry doesn't conflict with an Lua
+-- reserved word.  In the 'goto' example, with Lua 5.1 or earlier, 'goto' is
+-- not a reserved word so you'd be calling the Asterisk dialplan application
+-- 'goto'.  Lua 5.2 however, introduced the 'goto' control statement which
+-- makes 'goto' a reserved word.  This casues the interpreter to fail parsing
+-- the file and pbx_lua.so will fail to load.  The same applies to any use of
+-- Lua tables including extensions, channels and any tables you create.
+--
+-- There are two ways around this:  Since Lua is case-sensitive, you can use
+-- capitalized names, I.E. app.Goto(something) to refer to the Asterisk apps,
+-- functions, etc. Or you can use the full syntax, I.E. app["goto"](something).
+-- Both syntaxes are backwards compatible with earlier Lua versions.  To make
+-- your Lua dialplans easier to maintain and to reduce the chance of future
+-- conflicts you may want to use the app["goto"](something) syntax for all
+-- table accesses.
+--
+
+function outgoing_local(c, e)
+	app.dial("DAHDI/1/" .. e, "", "")
+end
+
+function demo_instruct()
+	app.background("demo-instruct")
+	app.waitexten()
+end
+
+function demo_congrats()
+	app.background("demo-congrats")
+	demo_instruct()
+end
+
+-- Answer the chanel and play the demo sound files
+function demo_start(context, exten)
+	app.wait(1)
+	app.answer()
+
+	channel.TIMEOUT("digit"):set(5)
+	channel.TIMEOUT("response"):set(10)
+	-- app.set("TIMEOUT(digit)=5")
+	-- app.set("TIMEOUT(response)=10")
+
+	demo_congrats(context, exten)
+end
+
+function demo_hangup()
+	app.playback("demo-thanks")
+	app.hangup()
+end
+
+extensions = {
+	demo = {
+		s = demo_start;
+
+		["2"] = function()
+			app.background("demo-moreinfo")
+			demo_instruct()
+		end;
+		["3"] = function ()
+			channel.LANGUAGE():set("fr") -- set the language to french
+			demo_congrats()
+		end;
+
+		["1000"] = function()
+--	See the naming conflict note above.
+			app['goto']("default", "s", 1)
+		end;
+
+		["1234"] = function()
+			app.playback("transfer", "skip")
+			-- do a dial here
+		end;
+
+		["1235"] = function()
+			app.voicemail("1234", "u")
+		end;
+
+		["1236"] = function()
+			app.dial("Console/dsp")
+			app.voicemail(1234, "b")
+		end;
+
+		["#"] = demo_hangup;
+		t = demo_hangup;
+                i = function()
+                        app.playback("invalid")
+                        demo_instruct()
+                end;
+
+		["500"] = function()
+			app.playback("demo-abouttotry")
+			app.dial("IAX2/guest@misery.digium.com/s@default")
+			app.playback("demo-nogo")
+			demo_instruct()
+		end;
+
+		["600"] = function()
+			app.playback("demo-echotest")
+			app.echo()
+			app.playback("demo-echodone")
+			demo_instruct()
+		end;
+
+		["8500"] = function()
+			app.voicemailmain()
+			demo_instruct()
+		end;
+
+	};
+
+	default = {
+		-- by default, do the demo
+		include = {"demo"};
+	};
+
+	public = {
+		-- ATTENTION: If your Asterisk is connected to the internet and you do
+		-- not have allowguest=no in sip.conf, everybody out there may use your
+		-- public context without authentication.  In that case you want to
+		-- double check which services you offer to the world.
+		--
+		include = {"demo"};
+	};
+
+	["local"] = {
+		["_NXXXXXX"] = outgoing_local;
+	};
+}
+
+hints = {
+	demo = {
+		[1000] = "SIP/1000";
+		[1001] = "SIP/1001";
+	};
+
+	default = {
+		["1234"] = "SIP/1234";
+	};
+}

version_1.8.2/pabx/etc/asterisk/extensions_anuncios.conf

@@ -0,0 +1,3 @@
+[ext-anuncios]
+include => ext-anuncios-custom
+exten => h,1,Macro(hangup)

version_1.8.2/pabx/etc/asterisk/extensions_callback.conf

@@ -0,0 +1,2 @@
+[ext-callback-externo]
+include => ext-callback-externo-custom

version_1.8.2/pabx/etc/asterisk/extensions_campanha.conf

@@ -0,0 +1 @@
+[ext-campanha]

version_1.8.2/pabx/etc/asterisk/extensions_conferencia.conf

@@ -0,0 +1,2 @@
+[ext-conferencia]
+include => ext-conferencia-custom

version_1.8.2/pabx/etc/asterisk/extensions_contextos_personalizados.conf

@@ -0,0 +1,33 @@
+[ext-ramais-custom]
+exten => _1100,1,ExecIf($["${ISNULL(${CDR(direcao)})}" = "1"]?Set(CDR(direcao)=I))
+exten => _1100,n,ExecIf($["${CDR(direcao)}" != "E" && "${ISNULL(${CDR(ramal_origem)})}" = "1"]?Set(CDR(ramal_origem)=${CALLERID(num)}))
+exten => _1100,n,Set(CHANNEL(accountcode)=${TRANSFER})
+exten => _1100,n,AGI(chefe-secretaria.php,${EXTEN},${CALLERID(num)})
+exten => _1100,n,GotoIf($["${TIPO}" = "EXTERNA"]?externa:interna)
+exten => _1100,n(interna),AGI(pbx_complemento.php,ext-ramais,interna)
+exten => _1100,n,Macro(dial-interno-manual,${EXTEN})
+exten => _1100,n,Hangup()
+exten => _1100,n(externa),AGI(pbx_complemento.php,ext-ramais)
+exten => _1100,n,Macro(dial-externo,${EXTEN})
+exten => _1100,n,Hangup()
+exten => _1101,1,ExecIf($["${ISNULL(${CDR(direcao)})}" = "1"]?Set(CDR(direcao)=I))
+exten => _1101,n,ExecIf($["${CDR(direcao)}" != "E" && "${ISNULL(${CDR(ramal_origem)})}" = "1"]?Set(CDR(ramal_origem)=${CALLERID(num)}))
+exten => _1101,n,Set(CHANNEL(accountcode)=${TRANSFER})
+exten => _1101,n,AGI(chefe-secretaria.php,${EXTEN},${CALLERID(num)})
+exten => _1101,n,GotoIf($["${TIPO}" = "EXTERNA"]?externa:interna)
+exten => _1101,n(interna),AGI(pbx_complemento.php,ext-ramais,interna)
+exten => _1101,n,Macro(dial-interno-manual,${EXTEN})
+exten => _1101,n,Hangup()
+exten => _1101,n(externa),AGI(pbx_complemento.php,ext-ramais)
+exten => _1101,n,Macro(dial-externo,${EXTEN})
+exten => _1101,n,Hangup()
+[ext-horarios-custom]
+[ext-fila-custom]
+[ext-disa-custom]
+[ext-grupos-custom]
+[ext-conferencia-custom]
+[ext-testes-custom]
+[saida-pstn-custom]
+[ext-anuncios-custom]
+[ext-callback-externo-custom]
+[ext-rotasinternas-custom]

version_1.8.2/pabx/etc/asterisk/extensions_disa.conf

@@ -0,0 +1,2 @@
+[ext-disa]
+include => ext-disa-custom

version_1.8.2/pabx/etc/asterisk/extensions_facilidades.conf

@@ -0,0 +1,369 @@
+[app-identifica-ramal]
+exten => *10,1,NoCDR()
+exten => *10,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *10,n,Playback(appsounds/ramal)
+exten => *10,n,SayDigits(${CALLERID(NUM)})
+exten => *10,n,Hangup()
+[app-hora]
+exten => *11,1,NoCDR()
+exten => *11,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *11,n,Answer()
+exten => *11,n,SayUnixTime(,,HMS)
+exten => *11,n,Hangup()
+[app-echo]
+exten => *12,1,NoCDR()
+exten => *12,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *12,n,Answer()
+exten => *12,n,Playback(demo-echotest)
+exten => *12,n,Echo()
+exten => *12,n,Playback(demo-echodone)
+exten => *12,n,Hangup()
+[app-troca-senha-ramal]
+exten => *13,1,NoCDR()
+exten => *13,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *13,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *13,n,Read(NOVASENHA,vm-newpassword,,n)
+exten => *13,n,GotoIf($["${LEN(${NOVASENHA})}" = "${LEN(${CALLERID(NUM)})}"]?confirmasenha)
+exten => *13,n,Playback(invalid)
+exten => *13,n,Hangup()
+exten => *13,n(confirmasenha),Read(CONFIRMASENHA,vm-reenterpassword,,n)
+exten => *13,n,GotoIf($["${NOVASENHA}" = "${CONFIRMASENHA}"]?final)
+exten => *13,n,Playback(incorrect-password)
+exten => *13,n,Hangup()
+exten => *13,n(final),Set(DB(SENHA/${CALLERID(NUM)})=${NOVASENHA})
+exten => *13,n,AGI(facilidades.php,${NOVASENHA})
+exten => *13,n,Wait(1)
+exten => *13,n,Playback(vm-passchanged&vm-goodbye)
+exten => *13,n,Hangup()
+[app-nao-pertube]
+exten => *14,1,NoCDR()
+exten => *14,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *14,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *14,n,Noop(Ativando função não pertube)
+exten => *14,n,Set(DB(NAOPERTUBE/${CALLERID(num)})=V)
+exten => *14,n,AGI(facilidades.php)
+exten => *14,n,Playback(beep&beep&activated)
+exten => *14,n,Hangup()
+exten => *15,1,NoCDR()
+exten => *15,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *15,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *15,n,Noop(Desativando função não pertube)
+exten => *15,n,Set(DB(NAOPERTUBE/${CALLERID(num)})=F)
+exten => *15,n,AGI(facilidades.php)
+exten => *15,n,Playback(beep&beep&de-activated)
+exten => *15,n,Hangup()
+[app-siga-me]
+exten => *17,1,NoCDR()
+exten => *17,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *17,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *17,n,Playback(vm-enter-num-to-call&vm-then-pound)
+exten => *17,n,Read(DESTINO,,,n) 
+exten => *17,n,Noop(Ativando função siga-me interno para o ramal ${DESTINO})
+exten => *17,n,Set(DB(SIGAME-INTERNO/${CALLERID(num)})=${DESTINO})
+exten => *17,n,AGI(facilidades.php,${DESTINO})
+exten => *17,n,Playback(beep&beep&activated)
+exten => *17,n,Hangup()
+exten => *18,1,NoCDR()
+exten => *18,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *18,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *18,n,Noop(Desativando função siga-me interno)
+exten => *18,n,Set(DB(SIGAME-INTERNO/${CALLERID(num)})=F)
+exten => *18,n,AGI(facilidades.php)
+exten => *18,n,Playback(beep&beep&de-activated)
+exten => *18,n,Hangup()
+exten => *19,1,NoCDR()
+exten => *19,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *19,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *19,n,Noop(Ativando função siga-me externo para o numero ${EXTEN:5})
+exten => *19,n,Playback(vm-enter-num-to-call&vm-then-pound)
+exten => *19,n,Read(DESTINO,,,n)
+exten => *19,n,Set(DB(SIGAME-EXTERNO/${CALLERID(num)})=${DESTINO})
+exten => *19,n,AGI(facilidades.php,${DESTINO})
+exten => *19,n,Playback(beep&beep&activated)
+exten => *19,n,Hangup()
+exten => *20,1,NoCDR()
+exten => *20,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *20,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *20,n,Noop(Desativando função siga-me externo)
+exten => *20,n,Set(DB(SIGAME-EXTERNO/${CALLERID(num)})=F)
+exten => *20,n,AGI(facilidades.php)
+exten => *20,n,Playback(beep&beep&de-activated)
+exten => *20,n,Hangup()
+exten => *21,1,NoCDR()
+exten => *21,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *21,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *21,n,Noop(Ativando função siga-me quando ocupado para o numero ${EXTEN:5})
+exten => *21,n,Playback(vm-enter-num-to-call&vm-then-pound)
+exten => *21,n,Read(DESTINO,,,n)
+exten => *21,n,Set(DB(SIGAME-OCUPADO/${CALLERID(num)})=${DESTINO})
+exten => *21,n,AGI(facilidades.php,${DESTINO})
+exten => *21,n,Playback(beep&beep&activated)
+exten => *21,n,Hangup()
+exten => *22,1,NoCDR()
+exten => *22,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *22,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *22,n,Noop(Desativando função siga-me ocupado)
+exten => *22,n,Set(DB(SIGAME-OCUPADO/${CALLERID(num)})=F)
+exten => *22,n,AGI(facilidades.php)
+exten => *22,n,Playback(beep&beep&de-activated)
+exten => *22,n,Hangup()
+exten => *23,1,NoCDR()
+exten => *23,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *23,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *23,n,Noop(Ativando função siga-me externo para o numero ${EXTEN:5})
+exten => *23,n(inicio),Playback(vm-enter-num-to-call&vm-then-pound)
+exten => *23,n,Read(DESTINO,,,n,,)
+exten => *23,n,GotoIf($["X${DESTINO}" = "X"]?invalido)
+exten => *23,n,Set(DB(SIGAME-NAOATENDE/${CALLERID(num)})=${DESTINO})
+exten => *23,n,AGI(facilidades.php,${DESTINO})
+exten => *23,n,Playback(beep&beep&activated)
+exten => *23,n,Hangup()
+exten => *23,n(invalido),Playback(invalid)
+exten => *23,n,Goto(inicio)
+exten => *24,1,NoCDR()
+exten => *24,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *24,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *24,n,Noop(Desativando função siga-me externo)
+exten => *24,n,Set(DB(SIGAME-NAOATENDE/${CALLERID(num)})=F)
+exten => *24,n,AGI(facilidades.php)
+exten => *24,n,Playback(beep&beep&de-activated)
+exten => *24,n,Hangup()
+[app-rechamada]
+exten => *25,1,NoCDR()
+exten => *25,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *25,n,Set(ARQUIVO=${DB(RECHAMADA/${CALLERID(NUM)})})
+exten => *25,n,System(/bin/rm ${ARQUIVO})
+exten => *25,n,Playback(beep&beep&activated)
+exten => *25,n,AGI(rechamada.php,${CALLERID(NUM)})
+exten => *25,n,Hangup()
+exten => *26,1,NoCDR()
+exten => *26,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *26,n,Set(ARQUIVO=${DB(RECHAMADA/${CALLERID(NUM)})})
+exten => *26,n,System(/bin/rm ${ARQUIVO})
+exten => *26,n,Set(DB(RECHAMADA/${CALLERID(NUM)})=F)
+exten => *26,n,Playback(beep&beep&de-activated)
+exten => *26,n,Hangup()
+[app-cadeado]
+exten => *27,1,NoCDR()
+exten => *27,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *27,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *27,n,Noop(Ativando função Cadeado Eletronico)
+exten => *27,n,Set(DB(CADEADO/${CALLERID(NUM)})=V)
+exten => *27,n,AGI(facilidades.php)
+exten => *27,n,Playback(beep&beep&activated)
+exten => *27,n,Hangup()
+exten => *28,1,NoCDR()
+exten => *28,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *28,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *28,n,Noop(Desativando função Cadeado Eletronico)
+exten => *28,n,Set(DB(CADEADO/${CALLERID(NUM)})=F)
+exten => *28,n,AGI(facilidades.php)
+exten => *28,n,Playback(beep&beep&de-activated)
+exten => *28,n,Hangup()
+[app-chamada-espera]
+exten => *29,1,NoCDR()
+exten => *29,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *29,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *29,n,Noop(Ativando função Cadeado Eletronico)
+exten => *29,n,Set(DB(CHAMADA-ESPERA/${CALLERID(NUM)})=V)
+exten => *29,n,AGI(facilidades.php)
+exten => *29,n,Playback(beep&beep&activated)
+exten => *29,n,Hangup()
+exten => *30,1,NoCDR()
+exten => *30,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *30,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *30,n,Noop(Desativando função Cadeado Eletronico)
+exten => *30,n,Set(DB(CHAMADA-ESPERA/${CALLERID(NUM)})=F)
+exten => *30,n,AGI(facilidades.php)
+exten => *30,n,Playback(beep&beep&de-activated)
+exten => *30,n,Hangup()
+[app-identifica-vm]
+exten => *32,1,NoCDR()
+exten => *32,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *32,n,Playback(appsounds/vm-caixapostal)
+exten => *32,n,Set(VM=${DB(VM/${CALLERID(NUM)})})
+exten => *32,n,SayDigits(${VM})
+exten => *32,n,Playback(auth-thankyou)
+exten => *32,n,Hangup()
+[app-acessa-vm]
+exten => *31,1,NoCDR()
+exten => *31,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *31,n,VoiceMailMain(${DB(VM/${CALLERID(NUM)})})
+exten => *31,n,Hangup()
+[app-agenda-ramal]
+exten => *33,1,NoCDR()
+exten => *33,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *33,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *33,n,Noop(Ativando Agenda Ramal)
+exten => *33,n(inicio),Background(speed-enterlocation&vm-then-pound,n)
+exten => *33,n,Read(POSICAO-AGENDA,,,n,,30)
+exten => *33,n,GotoIf($["X${POSICAO-AGENDA}" = "X"]?invalido)
+exten => *33,n,Background(speed-enternumber&vm-then-pound,n)
+exten => *33,n,Read(DESTINO-AGENDA,,,n,,30)
+exten => *33,n,GotoIf($["X${DESTINO-AGENDA}" = "X"]?invalido)
+exten => *33,n,Set(DB(AGENDA/${CALLERID(NUM)}/${POSICAO-AGENDA})=${DESTINO-AGENDA})
+exten => *33,n,AGI(facilidades.php,${CALLERID(NUM)},${POSICAO-AGENDA},${DESTINO-AGENDA})
+exten => *33,n,Playback(beep&beep&activated)
+exten => *33,n,Goto(inicio)
+exten => *33,n,Hangup()
+exten => *33,n(invalido),Playback(invalid)
+exten => *33,n,Goto(inicio)
+exten => *33,n,Hangup()
+exten => *34,1,NoCDR()
+exten => *34,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *34,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *34,n,Noop(Desativando Agenda Ramal)
+exten => *34,n(inicio),Background(speed-enterlocation&vm-then-pound,n)
+exten => *34,n,Read(POSICAO-AGENDA,,,n,,30)
+exten => *34,n,GotoIf($["X${POSICAO-AGENDA}" = "X"]?invalido)
+exten => *34,n,Noop(${DB_DELETE(AGENDA/${CALLERID(NUM)}/${POSICAO-AGENDA})})
+exten => *34,n,AGI(facilidades.php,${CALLERID(NUM)},${POSICAO-AGENDA},${DESTINO-AGENDA})
+exten => *34,n,Playback(beep&beep&de-activated)
+exten => *34,n,Goto(inicio)
+exten => *34,n,Hangup()
+exten => *34,n(invalido),Playback(invalid)
+exten => *34,n,Goto(inicio)
+exten => *34,n,Hangup()
+[app-agenda-publica]
+exten => *35,1,NoCDR()
+exten => *35,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *35,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *35,n,Noop(Ativando Agenda Publica)
+exten => *35,n(inicio),Background(speed-enterlocation&vm-then-pound,n)
+exten => *35,n,Read(POSICAO-AGENDA,,,n,,30)
+exten => *35,n,GotoIf($["X${POSICAO-AGENDA}" = "X"]?invalido)
+exten => *35,n,Background(speed-enternumber&vm-then-pound,n)
+exten => *35,n,Read(DESTINO-AGENDA,,,n,,30)
+exten => *35,n,GotoIf($["X${DESTINO-AGENDA}" = "X"]?invalido)
+exten => *35,n,Set(DB(AGENDA/${POSICAO-AGENDA})=${DESTINO-AGENDA})
+exten => *35,n,AGI(facilidades.php,${POSICAO-AGENDA},${DESTINO-AGENDA})
+exten => *35,n,Playback(beep&beep&activated)
+exten => *35,n,Goto(inicio)
+exten => *35,n,Hangup()
+exten => *35,n(invalido),Playback(invalid)
+exten => *35,n,Goto(inicio)
+exten => *35,n,Hangup()
+exten => *36,1,NoCDR()
+exten => *36,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *36,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *36,n,Noop(Desativando Agenda Publica)
+exten => *36,n(inicio),Background(speed-enterlocation&vm-then-pound,n)
+exten => *36,n,Read(POSICAO-AGENDA,,,n,,30)
+exten => *36,n,GotoIf($["X${POSICAO-AGENDA}" = "X"]?invalido)
+exten => *36,n,Noop(${DB_DELETE(AGENDA/${POSICAO-AGENDA})})
+exten => *36,n,AGI(facilidades.php,${POSICAO-AGENDA},${DESTINO-AGENDA})
+exten => *36,n,Playback(beep&beep&de-activated)
+exten => *36,n,Goto(inicio)
+exten => *36,n,Hangup()
+exten => *36,n(invalido),Playback(invalid)
+exten => *36,n,Goto(inicio)
+exten => *36,n,Hangup()
+[app-ultimo-recebido]
+exten => *16,1,NoCDR()
+exten => *16,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *16,n,SayDigits(${DB(ULTIMO-RECEBIDO/${CALLERID(num)})})
+exten => *16,n,Hangup()
+[app-monitoramento]
+exten => *37,1,NoCDR()
+exten => *37,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *37,n,GotoIf($["X${DB(MONITORAMENTO/${CALLERID(NUM)})}" = "X"]?negado)
+exten => *37,n(inicio),Read(DESTINO-MONITOR,vm-enter-num-to-call,,n,,)
+exten => *37,n,GotoIf($["X${DESTINO-MONITOR}" = "X"]?invalido)
+exten => *37,n,ChanSpy(${DB(DIAL/${DESTINO-MONITOR})},bq) 
+exten => *37,n,Hangup()
+exten => *37,n(invalido),Playback(invalid)
+exten => *37,n,Goto(inicio)
+exten => *37,n(negado),Playback(appsounds/ramal-invalido&vm-goodbye)
+exten => *37,n,Hangup()
+[app-intercalar]
+exten => *38,1,NoCDR()
+exten => *38,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *38,n,GotoIf($["X${DB(MONITORAMENTO/${CALLERID(NUM)})}" = "X"]?negado)
+exten => *38,n(inicio),Read(DESTINO-MONITOR,vm-enter-num-to-call,,n,,)
+exten => *38,n,GotoIf($["X${DESTINO-MONITOR}" = "X"]?invalido)
+exten => *38,n,ChanSpy(${DB(DIAL/${DESTINO-MONITOR})},bqw) 
+exten => *38,n,Hangup()
+exten => *38,n(invalido),Playback(invalid)
+exten => *38,n,Goto(inicio)
+exten => *38,n(negado),Playback(appsounds/ramal-invalido&vm-goodbye)
+exten => *38,n,Hangup()
+[app-discagem-agenda]
+exten => _*8.,1,AGI(pbx_complemento.php,app-discagem-agenda)
+exten => _*8.,n,GotoIf($["X${DB(AGENDA/${CALLERID(NUM)}/${EXTEN:2})}" = "X"]?vazia)
+exten => _*8.,n,Goto(saida-pstn,${DB(AGENDA/${CALLERID(NUM)}/${EXTEN:2})},disca)
+exten => _*8.,n,Hangup()
+exten => _*8.,n(vazia),Playback(speed-dial-empty)
+exten => _*8.,n,Hangup()
+exten => _*9.,1,AGI(pbx_complemento.php,app_discagem_agenda_ramal)
+exten => _*9.,n,GotoIf($["X${DB(AGENDA/${EXTEN:2})}" = "X"]?vazia)
+exten => _*9.,n,Goto(saida-pstn,${DB(AGENDA/${EXTEN:2})},disca)
+exten => _*9.,n,Hangup()
+exten => _*9.,n(vazia),Playback(speed-dial-empty)
+exten => _*9.,n,Hangup()
+[troca-senha]
+exten => *5,1,Answer()
+exten => *5,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *5,n,Goto(troca-senha-conta,s,1)
+[troca-senha-conta]
+exten => s,1,Answer()
+exten => s,n,Background(appsounds/codigodeacesso&beep)
+exten => s,n,WaitExten()
+exten => _XXXX,1,Set(__CONTA=${EXTEN})
+exten => _XXXX,n,Goto(troca-senha-senhaatual,s,1)
+[troca-senha-senhaatual]
+exten => s,1,Answer()
+exten => s,n,Background(appsounds/senhaatual&beep)
+exten => s,n,WaitExten()
+exten => _XXXX,1,Set(_ATUAL=${EXTEN})
+exten => _XXXX,n,AGI(confirma_senha.php)
+exten => _XXXX,n,Noop(atual ? = ${ATUAL})
+exten => _XXXX,n,Noop(atual no banco ? = ${ATUAL_DB})
+exten => _XXXX,n,GotoIf($["${ATUAL}" = "${ATUAL_DB}"]?novasenha:invalida)
+exten => _XXXX,n(invalida),Answer()
+exten => _XXXX,n,Playback(appsounds/senha-invalida)
+exten => _XXXX,n,Hangup()
+exten => _XXXX,n(novasenha),Goto(troca-senha-novasenha,s,1)
+[troca-senha-novasenha]
+exten => s,1,Answer()
+exten => s,n,Background(appsounds/novasenha&beep)
+exten => s,n,WaitExten()
+exten => _XXXX,1,Set(_NOVA=${EXTEN})
+exten => _XXXX,n,Goto(troca-senha-confirmasenha,s,1)
+[troca-senha-confirmasenha]
+exten => s,1,Answer()
+exten => s,n(confirme),Background(appsounds/confirme&beep)
+exten => s,n,WaitExten()
+exten => _XXXX,1,Set(_NOVA2=${EXTEN})
+exten => _XXXX,n,GotoIf($["${NOVA}" = "${NOVA2}"]?atualiza:invalida)
+exten => _XXXX,n(atualiza),Set(DATA=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
+exten => _XXXX,n,AGI(update.php)
+exten => _XXXX,n,Answer()
+exten => _XXXX,n,Playback(appsounds/alterada-sucesso)
+exten => _XXXX,n,Hangup()
+exten => _XXXX,n(invalida),Answer()
+exten => _XXXX,n,Playback(appsounds/senha-invalida)
+exten => _XXXX,n,Hangup()
+[captura-chamada]
+exten => _*76X.,1,AGI(pbx_complemento.php,aplicacoes)
+exten => _*76X.,n,PickUp(${EXTEN:3}@PICKUPMARK)
+exten => _*76X.,n,Hangup()
+[app-ausencia-grupo]
+exten => *40,1,NoCDR()
+exten => *40,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *40,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *40,n,Noop(Ativando funcao - AUSENCIA NO GRUPO)
+exten => *40,n,PauseQueueMember(,Local/${CALLERID(NUM)}@padrao/n)
+exten => *40,n,AGI(facilidades.php)
+exten => *40,n,Playback(beep&beep&activated)
+exten => *40,n,Hangup()
+exten => *41,1,NoCDR()
+exten => *41,n,AGI(pbx_complemento.php,aplicacoes)
+exten => *41,n,Authenticate(${DB(SENHA/${CALLERID(NUM)})},d,)
+exten => *41,n,Noop(Desativando funcao - AUSENCIA NO GRUPO)
+exten => *41,n,UnPauseQueueMember(,Local/${CALLERID(NUM)}@padrao/n)
+exten => *41,n,AGI(facilidades.php)
+exten => *41,n,Playback(beep&beep&de-activated)
+exten => *41,n,Hangup()
+[app-conferencia-dinamica]
+exten => *99,1,Macro(inicia-conferencia)
+exten => *99,n,Hangup()

version_1.8.2/pabx/etc/asterisk/extensions_fila.conf

@@ -0,0 +1 @@
+[ext-fila]

version_1.8.2/pabx/etc/asterisk/extensions_fila_login.conf

@@ -0,0 +1 @@
+[ext-fila-login]

version_1.8.2/pabx/etc/asterisk/extensions_general.conf

@@ -0,0 +1,23 @@
+clearglobalvars = yes
+static = yes
+writeprotect = no
+
+[globals]
+;Scripts CallCenter
+INSERIR = /var/lib/asterisk/scripts/callcenter/inserir-agente.php
+ATUALIZAR = /var/lib/asterisk/scripts/callcenter/atualizar-agente.php
+DELETAR = /var/lib/asterisk/scripts/callcenter/deletar-agente.php
+CANAL = /var/lib/asterisk/scripts/callcenter/atualizar-canal.php
+TRANSFERENCIA = /var/lib/asterisk/scripts/callcenter/atualizar-transferencia.php
+MONITORAR = /var/lib/asterisk/scripts/callcenter/atualizar-monitorar.php
+INTERCALAR = /var/lib/asterisk/scripts/callcenter/atualizar-intercalar.php
+DYNAMIC_FEATURES=app-hangup-agentes
+TRANSFER_CONTEXT=ext-transferencia
+TEMPODERING=45
+OPCOESDIAL=wWKk
+INTEGRACAO = /var/lib/asterisk/scripts/integracao/serverAgi.php
+INTEGRACAO_ABD = /var/lib/asterisk/scripts/integracao/abdAgi.php
+RECHAMADA = /var/lib/asterisk/scripts/rechamada.sh
+TIMEOUT_PREPAGO=86400000
+LOGOFFAGENTE = /var/lib/asterisk/scripts/callcenter/logoff-agente.php
+;LOGOFFAGENTERAMAL = /var/lib/asterisk/scripts/callcenter/logoff_agente_ramal.php

version_1.8.2/pabx/etc/asterisk/extensions_grupos.conf

@@ -0,0 +1,2 @@
+[ext-grupos]
+include => ext-grupos-custom

version_1.8.2/pabx/etc/asterisk/extensions_horarios.conf

@@ -0,0 +1,3 @@
+[ext-horarios]
+include => ext-horarios-custom
+exten => h,1,Macro(hangup)

version_1.8.2/pabx/etc/asterisk/extensions_minivm.conf

@@ -0,0 +1,158 @@
+; MINI-VOICEMAIL dialplan example
+; ---------------------------------------------------------------------------------------
+; ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+;
+;
+; This is an example on how to use the Mini-Voicemail system to build
+; voicemail systems.
+;
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; A macro to test the MINIVMACCOUNT dialplan function
+; 	Currently, accountcode and pincode is not used in the application
+; 	They where added to be used in dialplan scripting
+;
+;
+[macro-minivmfunctest]
+exten => s,1,set(account=${ARGV1})
+exten => minivm,n,verbose(1,-------------------- Minivm Function test - Accoutn ${account}  -------------)
+exten => s,n,verbose(1,---- Has account:       ${MINIVMACCOUNT(${account}:hasaccount)})
+exten => s,n,verbose(1,---- Fullname:          ${MINIVMACCOUNT(${account}:fullname)})
+exten => s,n,verbose(1,---- Email:             ${MINIVMACCOUNT(${account}:email)})
+exten => s,n,verbose(1,---- Pager:             ${MINIVMACCOUNT(${account}:pager)})
+exten => s,n,verbose(1,---- E-mail template:   ${MINIVMACCOUNT(${account}:etemplate)})
+exten => s,n,verbose(1,---- Pager template:    ${MINIVMACCOUNT(${account}:ptemplate)})
+exten => s,n,verbose(1,---- Account code:      ${MINIVMACCOUNT(${account}:accountcode)})
+exten => s,n,verbose(1,---- Path:              ${MINIVMACCOUNT(${account}:path)})
+exten => s,n,verbose(1,---- Pincode:           ${MINIVMACCOUNT(${account}:pincode)})
+exten => s,n,verbose(1,---- Time zone:         ${MINIVMACCOUNT(${account}:timezone)})
+exten => s,n,verbose(1,---- Language:          ${MINIVMACCOUNT(${account}:language)})
+; This requires setvar=customerclass=gold in the account configuration
+exten => s,n,verbose(1,---- Var:customerclass: ${MINIVMACCOUNT(${account}:customerclass)})
+
+[minivm-scenario1]
+; minivmtest tests the dialplan function MINIVMACCOUNT
+; Check the output in the console with verbose set
+exten => minivmtest,1,answer
+exten => minivmtest,n,wait(0.5)
+exten => minivmtest,n,set(ACCOUNT=do-not-spam-me@example.com)
+exten => minivmtest,n,macro(minivmfunctest, ${ACCOUNT})
+exten => minivmtest,n,playback(beep)
+exten => minivmtest,n,hangup
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; "minivm" tests a full scenario
+; Remember that users may hangup
+; This works both for users with accounts in minivm.conf and by just giving an e-mail address
+; without configuring an account
+exten => minivm,1,answer
+exten => minivm,n,wait(0.5)	; Wait for Voip channels to settle
+exten => minivm,n,set(account=oej@example.com)
+exten => minivm,n,noop(------------------------------------------- Minivm Greet   -------------)
+exten => minivm,n,minivmgreet(${account})
+exten => minivm,n,verbose(1,-- MINIVM_GREET_STATUS = ${MINIVM_GREET_STATUS} )
+exten => minivm,n,noop(------------------------------------------- Minivm Record  -------------)
+exten => minivm,n,minivmRecord(${account},b)
+exten => minivm,n,goto(minivmcleanup,1)
+
+; Cleanup after recording or hangup
+exten => minivmcleanup,1,noop(------------------------------------------- Minivm Notify  -------------)
+;Increment voicemail counter with 1. The counter will be used in the e-mail message
+;and in the filename
+exten => minivmcleanup,n,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?minivmrecordfailure,1)
+exten => minivmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
+exten => minivmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
+exten => minivmcleanup,n,minivmNotify(${account})
+exten => minivmcleanup,n,verbose(1,-- MINIVM_NOTIFY_STATUS = ${MINIVM_NOTIFY_STATUS} )
+; Now, clean up after sending voicemail
+exten => minivmcleanup,n,noop(------------------------------------------- Minivm Delete  -------------)
+exten => minivmcleanup,n,minivmdelete()
+exten => minivmcleanup,n,verbose(1,-- MINIVM_DELETE_STATUS = ${MINIVM_DELETE_STATUS} )
+
+;Recording failed
+exten => minivmrecordfailure,1,playback(vm-sorry)
+exten => minivmrecordfailure,n,wait(1)
+exten => minivmrecordfailure,n,hangup
+
+; If the user hangs up during the recording, we need to clean up
+; And send notifications
+exten => h,1,gotoif($[x${MINIVM_DELETE_STATUS} != x] ?h,stop)
+exten => h,n,noop(------------------------------------------- HANGUP during voicemail recording   -------------)
+exten => h,n,goto(minivmcleanup,1)
+exten => h,n(stop),noop(---Minivm DONE----)
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; Extension to record a greeting message
+; Call this like:
+;	macro(recordgreetings,alice@atlanta.example.com)
+;
+[macro-recordgreetings]
+exten => s,1,answer
+exten => s,n,wait(0.5)
+exten => s,n,set(account=${ARGV1])
+; This file give extra options not available here, needs to be edited
+; Change of password does not work
+exten => s,n(menu),background(vm-options)
+exten => 1,1,setvar(option=u)
+exten => 1,n,macro(minivmrec,${account},${option})
+exten => 1,n,goto(menu)
+exten => 2,1,setvar(option=b)
+exten => 2,n,macro(minivmrec,${account},${option})
+exten => 2,n,goto(menu)
+exten => 3,1,setvar(option=n)
+exten => 3,n,macro(minivmrec,${account},${option})
+exten => 3,n,goto(menu)
+exten => 4,1,setvar(option=t)
+exten => 4,n,macro(minivmrec,${account},${option})
+exten => 4,n,goto(menu)
+exten => *,1,playback(vm-thankyou)
+exten => *,n,wait(1)
+exten => *,n,hangup
+
+exten => i,1,playback(invalid)
+exten => i,n,goto(menu)
+
+[macro-minivmrec]
+exten => s,1,gotoif(${MINIVMACCOUNT(${account}:hasaccount)}?record)
+; Account is not configured in minivm.conf or realtime
+; Phony message, add something useful here
+exten => s,n,playback(privacy-incorrect)
+exten => s,n,macroreturn
+exten => record,1,minivmappmess(${ARGV1},${ARGV2})
+exten => record,n,noop(Recording status: ${MINIVM_APPMESS_STATUS})
+exten => record,n,macroreturn
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; To set a counter and use a template for voicemail to users without acounts
+; use something like this
+;
+; email address is in the "account" channel variable. Set from ast_db or a script
+; based on called ID
+
+exten => sendvoicemail,1,answer
+exten => sendvoicemail,n,wait(0.5)
+exten => sendvoicemail,n,set(domain=${CUT(account,@,2)})
+exten => sendvoicemail,n,set(country=${CUT(domain,.,2)})
+exten => sendvoicemail,n,minivmgreet(${account})
+exten => sendvoicemail,n,minivmRecord(${account},b)
+exten => sendvoicemail,n,goto(sendvmcleanup)
+
+exten => sendvmcleanup,1,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?done)
+; The counter is set in the domain directory, so we don't create one directory per user
+; The counter has the email in the name of the counter, increase it
+; Set the MVM_COUNTER variable that we use in the template
+exten => sendvmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
+exten => sendvmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
+; Increase a domain counter too, to see how many voicemails are sent to this domain
+; This is just for statistics
+exten => sendvmcleanup,n,set(MINIVMCOUNTER(${domain}:${domain}-all:inc) = 1)
+
+; Send voicemail in e-mail with country-specific template
+; The template need to be defined in minivm.conf
+;
+exten => sendvmcleanup,n,minivmNotify(${account}, ${country}_email)
+exten => sendvmcleanup,n,minivmDelete()
+
+exten => sendvmcleanup,n(done),wait(0.5)
+exten => sendvmcleanup,n,hangup
+
+exten => h,1,gotoif($[${MINIVM_RECORD_STATUS} = SUCCESS]?sendvmcleanup,1))