removido docker react para testes e implementado adaptacoes para video no sip general webrtc

version_1.8-reactNative/README.md

@@ -1,102 +0,0 @@
-
-
-# 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-reactNative/docker-compose.yml

@@ -1,32 +0,0 @@
-version: '3.7'
-networks:
-    working:
-        driver: bridge
-services:
-    react-app:
-        build:
-            context: pabx
-        restart: always
-        container_name: v1.8-reactNative
-        working_dir: /usr/src/
-        ports:
-            - 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
-        stdin_open: true
-        tty: true
-        networks:
-            - working
-        volumes:
-            - ./pabx/etc/asterisk/:/etc/asterisk/
-            - ./pabx/certificados/:/usr/src/certificates
-            - ${pasta_do_projeto}/asterisk/var_lib_asterisk/:/var/lib/asterisk/
-            - ${pasta_do_projeto}/asterisk/hdaux_utilitarios_scripts/:/hdaux/utilitarios/scripts
-

version_1.8-reactNative/env_example

@@ -1,26 +0,0 @@
-
-
-# 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-reactNative/pabx/Dockerfile

@@ -1,67 +0,0 @@
-FROM debian:bullseye
-
-RUN apt-get update && apt-get install -y \
-    build-essential \
-    vim \
-    htop \
-    wget \
-    libssl-dev \
-    libncurses5-dev \
-    uuid-dev \
-    libjansson-dev \
-    libxml2-dev \
-    sqlite3 \
-    libsqlite3-dev \
-    pkg-config \
-    subversion \
-    libsrtp2-dev \
-    curl \
-    git \
-    python3 \
-    python3-pip \
-    openjdk-17-jdk \
-    unzip \
-    lib32stdc++6 \
-    lib32z1 \
-    libglu1-mesa \
-    libopus-dev \
-    libvpx-dev \
-    xmlstarlet \
-    && rm -rf /var/lib/apt/lists/*
-
-WORKDIR /usr/src
-
-#Instalar o Asterisk
-RUN wget https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-13.38.3.tar.gz \
-    && tar xzf asterisk-13.38.3.tar.gz \
-    && rm asterisk-13.38.3.tar.gz \
-    && cd asterisk-13.38.3 \
-    && ./configure --with-pjproject-bundled --with-srtp=/usr/include/srtp2 \
-    && make menuselect.makeopts \
-    && menuselect/menuselect --disable chan_pjsip \
-                             --enable chan_sip \
-                             --enable cdr_csv \
-                             --enable func_cdr \
-                             --enable cdr_pgsql \
-                             --enable codec_opus \
-                             --enable res_format_attr_opus \
-                             --enable res_crypto \
-                             --enable res_http_websocket \
-                             --enable res_ari \
-                             --enable res_stasis \
-                             --enable res_stun_monitor \
-                             --enable res_phoneprov \
-                             --enable res_odbc \
-                             --enable res_srtp \
-                             menuselect.makeopts \
-    && make -j$(nproc) \
-    && make install \
-    && make config \
-    && ldconfig
-
-#Expor portas necessarias
-EXPOSE 8088
-
-ENTRYPOINT ["/usr/sbin/asterisk"]
-CMD ["-f"]
-

version_1.8-reactNative/pabx/build-asterisk_php.sh

@@ -1,48 +0,0 @@
-FROM debian:bullseye
-
-RUN apt-get update && apt-get install -y \
-    build-essential \
-    wget \
-    libssl-dev \
-    libncurses5-dev \
-    uuid-dev \
-    libjansson-dev \
-    libxml2-dev \
-    sqlite3 \
-    libsqlite3-dev \
-    pkg-config \
-    subversion \
-    libsrtp2-dev \  # Instalação da biblioteca libsrtp2-dev
-    && rm -rf /var/lib/apt/lists/*
-
-WORKDIR /usr/src
-
-RUN wget https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-13.38.3.tar.gz \
-    && tar xzf asterisk-13.38.3.tar.gz \
-    && rm asterisk-13.38.3.tar.gz \
-    && cd asterisk-13.38.3 \
-    && ./configure --libdir=/usr/lib64 --with-pjproject-bundled --with-srtp=/usr/include/srtp2
-    && make menuselect.makeopts \
-    && menuselect/menuselect --disable chan_pjsip \
-                             --enable chan_sip \
-                             --enable cdr_csv \
-                             --enable func_cdr \
-                             --enable cdr_pgsql \
-			     --enable codec_opus \
-			     --enable res_format_attr_opus \
-			     --enable res_crypto \
-			     --enable res_http_websocket \
-			     --enable res_ari \
-			     --enable res_stasis \
-			     --enable res_stun_monitor \
-			     --enable res_phoneprov \
-			     --enable res_odbc \
-			     --enable res_srtp \
-                             menuselect.makeopts \
-    && make -j$(nproc) \
-    && make install \
-    && make config \
-    && ldconfig
-
-CMD ["/usr/sbin/asterisk", "-f"]
-cd /usr/src/

version_1.8-reactNative/pabx/certificados/RootCA.crt

@@ -1,20 +0,0 @@
------BEGIN CERTIFICATE-----
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-yaKD/l5z2DOa0wfN9MXKaFY1mMWG2a35SYTZRvo5lsgpOe1oo+2PPlQPXdZsFHBG
-QmN8i9g=
------END CERTIFICATE-----

version_1.8-reactNative/pabx/certificados/RootCA.key

@@ -1,28 +0,0 @@
------BEGIN PRIVATE KEY-----
-MIIEvwIBADANBgkqhkiG9w0BAQEFAASCBKkwggSlAgEAAoIBAQDuzwVAB5KRM9eB
-7iuVqrT1Yvu2AC7FEG2pVNURieDLUgQvzmQpflbrhq7OJk8YpeFzXHUlI5Qg8RKD
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-PSbAALwrxsh7o6bLnyfK4fcgIQ==
------END PRIVATE KEY-----

version_1.8-reactNative/pabx/certificados/RootCA.pem

@@ -1,20 +0,0 @@
------BEGIN CERTIFICATE-----
-MIIDMTCCAhmgAwIBAgIUfwDYeruX8TSBfeO3MBY7/p6MuKEwDQYJKoZIhvcNAQEL
-BQAwKDELMAkGA1UEBhMCQlIxGTAXBgNVBAMMEFNpbXBsZXNpcC1XZWJydGMwHhcN
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------END CERTIFICATE-----

version_1.8-reactNative/pabx/certificados/RootCA.srl

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

version_1.8-reactNative/pabx/certificados/domains.ext

@@ -1,7 +0,0 @@
-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-reactNative/pabx/certificados/localhost.crt

@@ -1,21 +0,0 @@
------BEGIN CERTIFICATE-----
-MIIDjDCCAnSgAwIBAgIUNG5OJEfvNU5CgjYGFZ2KlO7AmYIwDQYJKoZIhvcNAQEL
-BQAwKDELMAkGA1UEBhMCQlIxGTAXBgNVBAMMEFNpbXBsZXNpcC1XZWJydGMwHhcN
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------END CERTIFICATE-----

version_1.8-reactNative/pabx/certificados/localhost.csr

@@ -1,17 +0,0 @@
------BEGIN CERTIFICATE REQUEST-----
-MIICuTCCAaECAQAwdDELMAkGA1UEBhMCQlIxEzARBgNVBAgMCk1hdG9Hcm9zc28x
-DzANBgNVBAcMBkN1aWFiYTElMCMGA1UECgwcU2ltcGxlc2lwLXdlYnJ0Yy1jZXJ0
-aWZpY2F0ZTEYMBYGA1UEAwwPbG9jYWxob3N0LmxvY2FsMIIBIjANBgkqhkiG9w0B
-AQEFAAOCAQ8AMIIBCgKCAQEAnc7jAQjR7gTUfY3E9WRc11CnP6tZBnv8iduUgQYK
-X4EbYC6DNRlfCzjSBdoFXku2ps6ey8c4XJvhthCQWHvRvjaLUIaRN7dO506eLUqf
-7G8KdefbZIaicA+J6LonOL8IjV6UrGyJFYVUEXUncLEtJxtoT1MPWjOT1kntvYQr
-hkTJmrKZqF0SJy2dUIMkZiONqBcSfG3Opjae7gTBejWzXyyyLYqvWY+ue++81447
-Qy98KZjf8fHmjdI6mhjx0MozI62xMq3znme9VQmtiLyPqnoa4AAFzOfNyq3OXP9E
-rqxF1hucSFfsrI1CgbMOQCnloP0/K07yNB3NaRQEbdvJ+wIDAQABoAAwDQYJKoZI
-hvcNAQELBQADggEBAEc7RJzFLFr/S/k7XV2CRaH4vRyrD3eo41EhkJtOkvle82bB
-A4zrGnmg5ss2jgyv4RPTPxc/6+FXD1sFD8PH0H9aBqrOD2HWEHV+AhBhD3/nY9P/
-PRcqcO0TS+6Xdft8GC9KHHGnwKROkrrgOAHj1JFdnmUVSi7kU0qkrqoNivXeqMry
-kvgJf6+arSxNJBTDzuAdjA/BmJs6ocIpnLXMo4T7zN02mt92cFDkqQZbS89EBSiD
-heyREeZZgauh04TbCgVf2nf98mBAupgBXFMAvT3i4+pa5S+brBZQNKQ1oDPC77at
-hhwyV8AksnlBk8rBX83TCxNGTqE1Q1ljfFgSpvQ=
------END CERTIFICATE REQUEST-----

version_1.8-reactNative/pabx/certificados/localhost.key

@@ -1,28 +0,0 @@
------BEGIN PRIVATE KEY-----
-MIIEvgIBADANBgkqhkiG9w0BAQEFAASCBKgwggSkAgEAAoIBAQCdzuMBCNHuBNR9
-jcT1ZFzXUKc/q1kGe/yJ25SBBgpfgRtgLoM1GV8LONIF2gVeS7amzp7Lxzhcm+G2
-EJBYe9G+NotQhpE3t07nTp4tSp/sbwp159tkhqJwD4nouic4vwiNXpSsbIkVhVQR
-dSdwsS0nG2hPUw9aM5PWSe29hCuGRMmaspmoXRInLZ1QgyRmI42oFxJ8bc6mNp7u
-BMF6NbNfLLItiq9Zj65777zXjjtDL3wpmN/x8eaN0jqaGPHQyjMjrbEyrfOeZ71V
-Ca2IvI+qehrgAAXM583Krc5c/0SurEXWG5xIV+ysjUKBsw5AKeWg/T8rTvI0Hc1p
-FARt28n7AgMBAAECggEABGCq12qBOFMV+e2m2w/Pq4FnQlLkg1OJK+HVesbKHuAb
-/E0OArkTc+I9oT/tCjKbfIEa0phSLe99EWngNzQoG1i0AKXa/WquW6hIpwT87rgd
-2Ct4YI+DjgxATR5s3BwghptrEGVqYBh0vo87atrfX8hY2mp5cI0hur4OREWYV38h
-18rILsATZHRSB4+h9oLHOPIcGpUV2Vr0mCUXU/OUdDLcCXBakB1U+npeacy1MKzL
-5sb+hK5W2Xu2qWwYMSmFzbpmxNuUPDES7LSD8mL7jiBcKWO4pnh7EMwqO8Sf1BfO
-B1smTSDymAd6QduvzhJPCzUOy3px/ZuSL/rNFLExAQKBgQDQtuM96Yh0OQX0lMbY
-i2TJVaR08DkmpDMKsxOFgyTh4mR+LWhpaBmOLD/0prwcFZl/ZMkwCdeMEUoAo5x2
-PFTaKXWSYgq632XQiojtQESICW/nKWhx0X0+MYsTmjjN0ZtBqPDPcr2id6wE+0T9
-hbr9UbnxxynrFBZ8umtC+ImROwKBgQDBj4PithBNT1tpGiDnkxt1yWOKlMTKBdfv
-CovsHhWMhQvCvnWiiG3OGkTrXtmWHLCpL2wXBz4vtDcY2jeJxBf+HfUF2ropSfKP
-UlOZSpdMeMdqYmbZzIG4QRgqA23O6YJ+zYeBCYctn+Nz+Q98WCEFYq5SGncOcKLN
-gb26fSseQQKBgQCe3IXMyyNkqz6gMlnD2x3ErV42793gTpzXKQTT28VH9YTeEXvm
-tyi0cXPgqMSHb91c/P6xhNw+ra4xHFdsG3HOeD4YRG28XWvmtnuQchl3ng9B2N+5
-umVk4Frmeb2A02kkKg3ZKJG0xvf09rhFa/Ou+vTYVerPFoQjPdWErR4wnwKBgQCA
-WSrp7TIYIZQKuHGzPP/LdfQUk5t+HqvV7VdlBXniiOKIiwSz9uVg8lha/rUnbfTS
-6wgwqPPabBTQ3PpQ1v/V2IV+EVC6SjeWQ8/9aNG0uq345DExIQOvazKbwVRSDNMk
-RCMjNG83mLsZTsCvAj07NflDphrmL1g1D2AAxjkOQQKBgCDXz/EXXMDpEU02kYpV
-ovVGQ0qUH2oZ+ATax34hNvsLEuzENkcUFzIqQjXxoxwylddFvaIhK0KWACy3WJDe
-rhp89NJjvycKrzlYbs8sd5BdGfeqAKhXQxF8H4w+BBShRKzMA1iBr07hsn2gA1H7
-tU9hcLllm+PQpHOHb5T9gR8P
------END PRIVATE KEY-----

version_1.8-reactNative/pabx/config-docker.php

@@ -1,18 +0,0 @@
-<?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-reactNative/pabx/config_network

@@ -1,40 +0,0 @@
-#!/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.conf )"  ]]
-then
-	sed -i "s/;\?externaddr=.*/externaddr=${IP_HOST}/g" /etc/asterisk/sip_general.conf
-else
-	echo "externaddr=${IP_HOST}" >> /etc/asterisk/sip_general.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.conf )" ]]
-then
-	sed -i "s/;\?localnet=.*/localnet=0.0.0.0\/32/g" /etc/asterisk/sip_general.conf
-else
-	echo "localnet=0.0.0.0/32" >> /etc/asterisk/sip_general.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.conf)" ]]
-then
-	sed -i "s/;\?media_address=.*/media_address=${IP_HOST}/g" /etc/asterisk/sip_general.conf
-else
-	echo "media_address=${IP_HOST}" >> /etc/astersk/sip_general.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-reactNative/pabx/crontab.sh

@@ -1,40 +0,0 @@
-#!/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-reactNative/pabx/docker-entrypoint.sh

@@ -1,31 +0,0 @@
-#!/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-reactNative/pabx/etc/apache2/sites-available/pabx_simples.conf

@@ -1,100 +0,0 @@
-<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-reactNative/pabx/etc/asterisk/.asterisk.conf.swp:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/.extensions.conf.swk:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/.extensions.conf.swl:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/.extensions.conf.swm:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/.extensions.conf.swn:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/.extensions.conf.swo:Zone.Identifier

@@ -1,3 +0,0 @@
-[ZoneTransfer]
-ZoneId=3
-ReferrerUrl=\\wsl.localhost\Debian\home\matheorb\docker-pabx-app\version_1.8-reactNative\pabx\etc\etc_asterisk.tar.gz

version_1.8-reactNative/pabx/etc/asterisk/acl.conf

@@ -1,80 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/adsi.conf

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

version_1.8-reactNative/pabx/etc/asterisk/agents.conf

@@ -1,70 +0,0 @@
-;
-; Agent pool configuration
-;
-
-[general]
-; The general section of this config is not currently used, but reserved
-; for future use.
-
-;[agent-id]
-; Define ackcall to require the agent to give a DTMF acknowledgement
-; when the agent receives a call.
-; The channel variable AGENTACKCALL overrides on agent login.
-; Default is "no".
-;ackcall=no
-;
-; Set what DTMF key sequence the agent should use to acknowledge a call.
-; The channel variable AGENTACCEPTDTMF overrides on agent login.
-; This option is ignored unless ackcall is enabled.
-; Default is "#".
-;acceptdtmf=##
-;
-; Set how many seconds a call for the agent has to wait for the agent to
-; acknowledge the call before the agent is automatically logged off.  If
-; set to zero then the call will wait forever for the agent to acknowledge.
-; The channel variable AGENTAUTOLOGOFF overrides on agent login.
-; This option is ignored unless ackcall is enabled.
-; Default is 0.
-;autologoff=15
-;
-; Set the minimum amount of time after disconnecting a call before
-; the agent can receive a new call in milliseconds.
-; The channel variable AGENTWRAPUPTIME overrides on agent login.
-; Default is 0.
-;wrapuptime=5000
-;
-; Set the musiconhold class for the agent.
-; Default is "default".
-;musiconhold=default
-;
-; Enable recording calls the agent takes automatically by invoking the
-; DTMF automixmon feature when the agent connects to a caller.
-; See features.conf.sample for information about the automixmon feature.
-; Default is "no".
-;recordagentcalls=yes
-;
-; The sound file played to alert the agent when a call is present.
-; Default is "beep".
-;custom_beep=beep
-;
-; A friendly name for the agent used in log messages.
-; Default is "".
-;fullname=Mark Spencer
-;
-; --------------------------------------------------
-;
-; This section contains example agent definitions:
-;
-; Define a template called my-agents:
-;[my-agents](!)
-;autologoff=15
-;ackcall=yes
-;acceptdtmf=##
-;
-; Define agent 1001 using the my-agents template:
-;[1001](my-agents)
-;fullname=Mark Spencer
-;
-; Define agent 1002 using the my-agents template:
-;[1002](my-agents)
-;fullname=Will Meadows

version_1.8-reactNative/pabx/etc/asterisk/alarmreceiver.conf

@@ -1,91 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/alsa.conf

@@ -1,77 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/amd.conf

@@ -1,31 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/app_mysql.conf

@@ -1,24 +0,0 @@
-; 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-reactNative/pabx/etc/asterisk/app_skel.conf

@@ -1,27 +0,0 @@
-[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-reactNative/pabx/etc/asterisk/ari.conf

@@ -1,30 +0,0 @@
-[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-reactNative/pabx/etc/asterisk/ast_debug_tools.conf

@@ -1,63 +0,0 @@
-#
-# 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))
-
-# The directory to contain output files and work directories.
-# For output from existing core files, the default is the
-# directory that the core file is found in.  For core files
-# produced from a running process, the default is /tmp.
-OUTPUTDIR=/tmp
-
-# 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-reactNative/pabx/etc/asterisk/asterisk.adsi

@@ -1,158 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/asterisk.conf

@@ -1,123 +0,0 @@
-[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 = 3
-;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_media_frames = yes	; Cache media frames for performance
-				; Disable this option to help track down media frame
-				; mismanagement when using valgrind or MALLOC_DEBUG.
-				; The cache gets in the way of determining if the
-				; frame is used after being freed and who freed it.
-				; NOTE: This option has no effect when Asterisk is
-				; compiled with the LOW_MEMORY compile time option
-				; enabled because the cache code does not exist.
-				; Default yes
-;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 = asterisk		; The user to run as.
-;rungroup = asterisk		; 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-reactNative/pabx/etc/asterisk/calendar.conf

@@ -1,109 +0,0 @@
-;[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-reactNative/pabx/etc/asterisk/ccss.conf

@@ -1,205 +0,0 @@
-;
-; --- 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-reactNative/pabx/etc/asterisk/cdr.conf

@@ -1,166 +0,0 @@
-;
-; 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
-
-; 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-reactNative/pabx/etc/asterisk/cdr_adaptive_odbc.conf

@@ -1,59 +0,0 @@
-; 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-reactNative/pabx/etc/asterisk/cdr_custom.conf

@@ -1,14 +0,0 @@
-;
-; 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)}
-;
-; 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-reactNative/pabx/etc/asterisk/cdr_manager.conf

@@ -1,15 +0,0 @@
-;
-; Asterisk Call Management CDR
-;
-[general]
-enabled = no
-
-; The "mappings" category can be used to define additional "key: value" pairs
-; that will be included in the manager event. (after AccountCode, Source, etc).
-;
-; Each line like "varname => label" will include a "label: ${CDR(varname)}"
-; in the generated event where ${CDR(varname)} its replaced with its value
-;
-;[mappings]
-;rate => Rate
-;carrier => Carrier

version_1.8-reactNative/pabx/etc/asterisk/cdr_mysql.conf

@@ -1,62 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cdr_odbc.conf

@@ -1,12 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cdr_pgsql.conf

@@ -1,16 +0,0 @@
-; Sample Asterisk config file for CDR logging to PostgresSQL
-;
-; Note that you can use TLS connections to your database server.
-; This is configured for your PostgreSQL client installation
-; on this system (check for pgsessions.conf)
-
-[global]
-;hostname=localhost
-;port=5432
-;dbname=asterisk
-;password=password
-;user=postgres
-;appname=asterisk    ; Postgres application_name support (optional). Whitespace not allowed.
-;table=cdr		;SQL table where CDRs will be inserted
-;encoding=LATIN9	; Encoding of logged characters in Asterisk
-;timezone=UTC		; Uncomment if you want datetime fields in UTC/GMT

version_1.8-reactNative/pabx/etc/asterisk/cdr_sqlite3_custom.conf

@@ -1,10 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cdr_syslog.conf

@@ -1,83 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cdr_tds.conf

@@ -1,77 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cel.conf

@@ -1,116 +0,0 @@
-;
-; 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=yes
-
-; 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-reactNative/pabx/etc/asterisk/cel_custom.conf

@@ -1,35 +0,0 @@
-;
-; 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})}

version_1.8-reactNative/pabx/etc/asterisk/cel_odbc.conf

@@ -1,108 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cel_pgsql.conf

@@ -1,68 +0,0 @@
-;
-; 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
-
-[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=localhost
-;port=5432
-;dbname=asterisk
-;password=password
-;user=postgres
-;table=cel		;SQL table where CEL's will be inserted
-;appname=asterisk   ; Postgres application_name support (optional). Whitespace not allowed.

version_1.8-reactNative/pabx/etc/asterisk/cel_sqlite3_custom.conf

@@ -1,24 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cel_tds.conf

@@ -1,69 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/chan_mobile.conf

@@ -1,69 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cli.conf

@@ -1,12 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cli_aliases.conf

@@ -1,203 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/cli_permissions.conf

@@ -1,82 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/codecs.conf

@@ -1,211 +0,0 @@
-[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.
-; By default plc is applied only when the 2 codecs
-; in a channel are different.
-genericplc => true
-; Apply generic plc to channels even if the 2 codecs
-; are the same.  This forces transcoding via slin so
-; the performance impact should be considered.
-; Ignored if genericplc is not also enabled.
-genericplc_on_equal_codecs => false
-
-; 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 ============================
-;
-; NOTE: Accurate documentation corresponding to your downloaded version of
-;       codec_opus is available from Asterisk's CLI:
-;
-;       *CLI> config show help codec_opus opus
-;
-;[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. (default: 48000)
-;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. (default: auto)
-;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. (default: no)
-;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.
-      ; (default: yes)
-;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. (default: no)
-
-;=============================== OPUS Examples ================================
-;
-;[opus]
-;type=opus
-;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder
-;fec=no ; No inband fec
-
-;[myopus]
-;type=opus
-;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz
-;                   ; audio bandwidth at 16Khz sample rate)
-;cbr=yes ; Negotiate a constant bit rate

version_1.8-reactNative/pabx/etc/asterisk/confbridge.conf

@@ -1,395 +0,0 @@
-[general]
-; The general section of this config
-; is not currently used, but reserved
-; for future use.
-
-;
-; --- Default Information ---
-; The default_user and default_bridge sections are applied
-; automatically to all ConfBridge instances invoked without
-; a user, or bridge argument.  No menu is applied by default.
-;
-; Note that while properties of the default_user or default_bridge
-; profile can be overridden, if removed, they will be automatically
-; added and made available to the dialplan upon module load.
-;
-
-; --- ConfBridge User Profile Options ---
-[default_user]
-type=user
-;admin=yes     ; Sets if the user is an admin or not. Off by default.
-;marked=yes    ; Sets if this is a marked user or not. Off by default.
-;startmuted=yes; Sets if all users should start out muted. Off by default
-;music_on_hold_when_empty=yes  ; Sets whether MOH should be played when only
-                               ; one person is in the conference or when the
-                               ; the user is waiting on a marked user to enter
-                               ; the conference. Off by default.
-;music_on_hold_class=default   ; The MOH class to use for this user.
-;quiet=yes     ; When enabled enter/leave prompts and user intros are not played.
-               ; There are some prompts, such as the prompt to enter a PIN number,
-               ; that must be played regardless of what this option is set to.
-               ; Off by default
-;announce_user_count=yes  ; Sets if the number of users should be announced to the
-                          ; caller.  Off by default.
-;announce_user_count_all=yes ; Sets if the number of users should be announced to
-                             ; all the other users in the conference when someone joins.
-                             ; This option can be either set to 'yes' or a number.
-                             ; When set to a number, the announcement will only occur
-                             ; once the user count is above the specified number.
-;announce_only_user=yes   ; Sets if the only user announcement should be played
-                          ; when a channel enters a empty conference.  On by default.
-;wait_marked=yes   ; Sets if the user must wait for a marked user to enter before
-                   ; joining the conference. Off by default.
-;end_marked=yes ; This option will kick every user with this option set in their
-                ; user profile after the last Marked user exists the conference.
-
-;dsp_drop_silence=yes  ; This option drops what Asterisk detects as silence from
-                       ; entering into the bridge.  Enabling this option will drastically
-                       ; improve performance and help remove the buildup of background
-                       ; noise from the conference. Highly recommended for large conferences
-                       ; due to its performance enhancements.
-
-;dsp_talking_threshold=128  ; Average magnitude threshold to determine talking.
-                            ;
-                            ; The minimum average magnitude per sample in a frame for the
-                            ; DSP to consider talking/noise present.  A value below this
-                            ; level is considered silence.  This value affects several
-                            ; operations and should not be changed unless the impact on
-                            ; call quality is fully understood.
-                            ;
-                            ; What this value affects internally:
-                            ;
-                            ; 1. Audio is only mixed out of a user's incoming audio
-                            ;    stream if talking is detected.  If this value is set too
-                            ;    high the user will hear himself talking.
-                            ;
-                            ; 2. When talk detection AMI events are enabled, this value
-                            ;    determines when talking has begun which results in an
-                            ;    AMI event to fire.  If this value is set too low AMI
-                            ;    events may be falsely triggered by variants in room
-                            ;    noise.
-                            ;
-                            ; 3. The 'drop_silence' option depends on this value to
-                            ;    determine when the user's audio should be mixed into the
-                            ;    bridge after periods of silence.  If this value is too
-                            ;    high the user's speech will get discarded as they will
-                            ;    be considered silent.
-                            ;
-                            ; Valid values are 1 through 2^15.
-                            ; By default this value is 160.
-
-;dsp_silence_threshold=2000 ; The number of milliseconds of silence necessary to declare
-                            ; talking stopped.
-                            ;
-                            ; The time in milliseconds of sound falling below the
-                            ; 'dsp_talking_threshold' option when a user is considered to
-                            ; stop talking.  This value affects several operations and
-                            ; should not be changed unless the impact on call quality is
-                            ; fully understood.
-                            ;
-                            ; What this value affects internally:
-                            ;
-                            ; 1. When talk detection AMI events are enabled, this value
-                            ;    determines when the user has stopped talking after a
-                            ;    period of talking.  If this value is set too low AMI
-                            ;    events indicating the user has stopped talking may get
-                            ;    falsely sent out when the user briefly pauses during mid
-                            ;    sentence.
-                            ;
-                            ; 2. The 'drop_silence' option depends on this value to
-                            ;    determine when the user's audio should begin to be
-                            ;    dropped from the conference bridge after the user stops
-                            ;    talking.  If this value is set too low the user's audio
-                            ;    stream may sound choppy to the other participants.  This
-                            ;    is caused by the user transitioning constantly from
-                            ;    silence to talking during mid sentence.
-                            ;
-                            ; The best way to approach this option is to set it slightly
-                            ; above the maximum amount of milliseconds of silence a user
-                            ; may generate during natural speech.
-                            ;
-                            ; Valid values are 1 through 2^31.
-                            ; By default this value is 2500ms.
-
-;talk_detection_events=yes ; This option sets whether or not notifications of when a user
-                           ; begins and ends talking should be sent out as events over AMI.
-                           ; By default this option is off.
-
-;denoise=yes ; Sets whether or not a denoise filter should be applied
-             ; to the audio before mixing or not.  Off by default. Requires
-             ; func_speex to be built and installed.  Do not confuse this option
-             ; with drop_silence.  Denoise is useful if there is a lot of background
-             ; noise for a user as it attempts to remove the noise while preserving
-             ; the speech.  This option does NOT remove silence from being mixed into
-             ; the conference and does come at the cost of a slight performance hit.
-
-;jitterbuffer=yes  ; Enabling this option places a jitterbuffer on the user's audio stream
-                   ; before audio mixing is performed.  This is highly recommended but will
-                   ; add a slight delay to the audio.  This option is using the JITTERBUFFER
-                   ; dialplan function's default adaptive jitterbuffer.  For a more fine tuned
-                   ; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
-                   ; on the user before entering the ConfBridge application.
-
-;pin=1234  ; Sets if this user must enter a PIN number before entering
-           ; the conference.  The PIN will be prompted for.
-;announce_join_leave=yes ; When enabled, this option will prompt the user for a
-                         ; name when entering the conference.  After the name is
-                         ; recorded, it will be played as the user enters and exists
-                         ; the conference. This option is off by default.
-;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
-                                ; will be prompted to review their recording before
-                                ; entering the conference. During this phase, the recording
-                                ; may be listened to, re-recorded, or accepted as is. This
-                                ; option is off by default.
-;dtmf_passthrough=yes  ; Sets whether or not DTMF should pass through the conference.
-                       ; This option is off by default.
-;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
-
-;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
-              ; may stay in the conference before being automatically ejected. When the user
-              ; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
-              ; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
-              ; Default: 0
-
-; --- ConfBridge Bridge Profile Options ---
-[default_bridge]
-type=bridge
-;max_members=50                ; This option limits the number of participants for a single
-                               ; conference to a specific number.  By default conferences
-                               ; have no participant limit. After the limit is reached, the
-                               ; conference will be locked until someone leaves.  Note however
-                               ; that an Admin user will always be alowed to join the conference
-                               ; regardless if this limit is reached or not.
-
-;record_conference=yes         ; Records the conference call starting when the first user
-                               ; enters the room, and ending when the last user exits the room.
-                               ; The default recorded filename is
-                               ; 'confbridge-<name of conference bridge>-<start time>.wav
-                               ; and the default format is 8khz slinear.  This file will be
-                               ; located in the configured monitoring directory in asterisk.conf.
-
-;record_file=</path/to/file>   ; When record_conference is set to yes, the specific name of the
-                               ; record file can be set using this option.  Note that since multiple
-                               ; conferences may use the same bridge profile, this may cause issues
-                               ; depending on the configuration.  It is recommended to only use this
-                               ; option dynamically with the CONFBRIDGE() dialplan function. This
-                               ; allows the record name to be specified and a unique name to be chosen.
-                               ; By default, the record_file is stored in Asterisk's spool/monitor directory
-                               ; with a unique filename starting with the 'confbridge' prefix.
-
-;internal_sample_rate=auto     ; Sets the internal native sample rate the
-                               ; conference is mixed at.  This is set to automatically
-                               ; adjust the sample rate to the best quality by default.
-                               ; Other values can be anything from 8000-192000.  If a
-                               ; sample rate is set that Asterisk does not support, the
-                               ; closest sample rate Asterisk does support to the one requested
-                               ; will be used.
-
-;maximum_sample_rate=none      ; Sets the maximum sample rate the conference
-                               ; is mixed at. This is set to no maximum by default.
-                               ; Values can be anything from 8000-192000.
-
-;mixing_interval=40     ; Sets the internal mixing interval in milliseconds for the bridge.  This
-                        ; number reflects how tight or loose the mixing will be for the conference.
-                        ; In order to improve performance a larger mixing interval such as 40ms may
-                        ; be chosen.  Using a larger mixing interval comes at the cost of introducing
-                        ; larger amounts of delay into the bridge.  Valid values here are 10, 20, 40,
-                        ; or 80.  By default 20ms is used.
-
-;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
-                           ; Note that participants wanting to view and be the source of a video feed
-                           ; _MUST_ be sharing the same video codec.  Also, using video in conjunction with
-                           ; with the jitterbuffer currently results in the audio being slightly out of sync
-                           ; with the video.  This is a result of the jitterbuffer only working on the audio
-                           ; stream.  It is recommended to disable the jitterbuffer when video is used.
-                           ;
-                           ; --- MODES ---
-                           ; none: No video sources are set by default in the conference. It is still
-                           ;       possible for a user to be set as a video source via AMI or DTMF action
-                           ;       at any time.
-                           ;
-                           ; follow_talker: The video feed will follow whoever is talking and providing video.
-                           ;
-                           ; last_marked: The last marked user to join the conference with video capabilities
-                           ;              will be the single source of video distributed to all participants.
-                           ;              If multiple marked users are capable of video, the last one to join
-                           ;              is always the source, when that user leaves it goes to the one who
-                           ;              joined before them.
-                           ;
-                           ; first_marked: The first marked user to join the conference with video capabilities
-                           ;               is the single source of video distribution among all participants. If
-                           ;               that user leaves, the marked user to join after them becomes the source.
-
-;language=en               ; Set the language used for announcements to the conference.
-                           ; Default is en (English).
-
-;regcontext=conferences    ; The name of the context into which to register conference names as extensions.
-
-; All sounds in the conference are customizable using the bridge profile options below.
-; Simply state the option followed by the filename or full path of the filename after
-; the option.  Example: sound_had_joined=conf-hasjoin  This will play the conf-hasjoin
-; sound file found in the sounds directory when announcing someone's name is joining the
-; conference.
-
-;sound_join  ; The sound played to everyone when someone enters the conference.
-;sound_leave ; The sound played to everyone when someone leaves the conference.
-;sound_has_joined ; The sound played before announcing someone's name has
-                  ; joined the conference. This is used for user intros.
-                  ; Example "_____ has joined the conference"
-;sound_has_left ; The sound played when announcing someone's name has
-                ; left the conference. This is used for user intros.
-                ; Example "_____ has left the conference"
-;sound_kicked ; The sound played to a user who has been kicked from the conference.
-;sound_muted  ; The sound played when the mute option is toggled on using DTMF menu.
-;sound_unmuted  ; The sound played when the mute option is toggled off using DTMF menu.
-;sound_only_person ; The sound played when the user is the only person in the conference.
-;sound_only_one ; The sound played to a user when there is only one other
-                ; person is in the conference.
-;sound_there_are  ; The sound played when announcing how many users there
-                  ; are in a conference.
-;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
-                       ; when announcing how many users there are in the conference.
-                       ; The sounds are stringed together like this.
-                       ; "sound_there_are" <number of participants> "sound_other_in_party"
-;sound_place_into_conference ; The sound played when someone is placed into the conference
-                             ; after waiting for a marked user. This sound is now deprecated
-                             ; since it was only ever used improperly and correcting that bug
-                             ; made it completely unused.
-;sound_wait_for_leader  ; The sound played when a user is placed into a conference that
-                        ; can not start until a marked user enters.
-;sound_leader_has_left  ; The sound played when the last marked user leaves the conference.
-;sound_get_pin ; The sound played when prompting for a conference pin number.
-;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
-;sound_locked ; The sound played to a user trying to join a locked conference.
-;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
-;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
-;sound_error_menu ; The sound played when an invalid menu option is entered.
-;sound_begin ; The sound played to the conference when the first marked user enters the conference.
-
-; --- ConfBridge Menu Options ---
-; The ConfBridge application also has the ability to
-; apply custom DTMF menus to each channel using the
-; application.  Like the User and Bridge profiles
-; a menu is passed in to ConfBridge as an argument in
-; the dialplan.
-;
-; Below is a list of menu actions that can be assigned
-; to a DTMF sequence.
-;
-; To have the first DTMF digit in a sequence be the '#' character, you need to
-; escape it.  If it is not escaped then normal config file processing will
-; think it is a directive like #include.  For example:
-; \#1=toggle_mute                      ; Pressing #1 will toggle the mute setting.
-;
-; A single DTMF sequence can have multiple actions associated with it. This is
-; accomplished by stringing the actions together and using a ',' as the delimiter.
-; Example:  Both listening and talking volume is reset when '5' is pressed.
-; 5=reset_talking_volume, reset_listening_volume
-;
-; playback(<name of audio file>&<name of audio file>)
-                                       ; Playback will play back an audio file to a channel
-                                       ; and then immediately return to the conference.
-                                       ; This file can not be interupted by DTMF.
-                                       ; Mutliple files can be chained together using the
-                                       ; '&' character.
-; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
-                                       ; playback_and_continue will
-                                       ; play back a prompt while continuing to
-                                       ; collect the dtmf sequence.  This is useful
-                                       ; when using a menu prompt that describes all
-                                       ; the menu options.  Note however that any DTMF
-                                       ; during this action will terminate the prompts
-                                       ; playback.  Prompt files can be chained together
-                                       ; using the '&' character as a delimiter.
-; toggle_mute      ; Toggle turning on and off mute.  Mute will make the user silent
-                   ; to everyone else, but the user will still be able to listen in.
-
-; no_op ; This action does nothing (No Operation). Its only real purpose exists for
-        ; being able to reserve a sequence in the config as a menu exit sequence.
-; decrease_listening_volume ; Decreases the channel's listening volume.
-; increase_listening_volume ; Increases the channel's listening volume.
-; reset_listening_volume    ; Reset channel's listening volume to default level.
-
-; decrease_talking_volume ; Decreases the channel's talking volume.
-; increase_talking_volume ; Icreases the channel's talking volume.
-; reset_talking_volume    ; Reset channel's talking volume to default level.
-;
-; dialplan_exec(context,exten,priority)  ; The dialplan_exec action allows a user
-                                         ; to escape from the conference and execute
-                                         ; commands in the dialplan.  Once the dialplan
-                                         ; exits the user will be put back into the
-                                         ; conference.  The possibilities are endless!
-; leave_conference ; This action allows a user to exit the conference and continue
-                   ; execution in the dialplan.
-;
-; admin_kick_last  ; This action allows an Admin to kick the last participant from the
-                   ; conference. This action will only work for admins which allows
-                   ; a single menu to be used for both users and admins.
-;
-; admin_toggle_conference_lock ; This action allows an Admin to toggle locking and
-                               ; unlocking the conference.  Non admins can not use
-                               ; this action even if it is in their menu.
-
-; set_as_single_video_src   ; This action allows any user to set themselves as the
-                            ; single video source distributed to all participants.
-                            ; This will make the video feed stick to them regardless
-                            ; of what the video_mode is set to.
-
-; release_as_single_video_src ; This action allows a user to release themselves as
-                              ; the video source.  If video_mode is not set to "none"
-                              ; this action will result in the conference returning to
-                              ; whatever video mode the bridge profile is using.
-                              ;
-                              ; Note that this action will have no effect if the user
-                              ; is not currently the video source.  Also, the user is
-                              ; not guaranteed by using this action that they will not
-                              ; become the video source again.  The bridge will return
-                              ; to whatever operation the video_mode option is set to
-                              ; upon release of the video src.
-
-; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
-                                 ; state for all non-admins within a conference.
-                                 ; Subsequent non-admins joining a muted conference will
-                                 ; start muted.  All admin users are unaffected by this
-                                 ; option.  Note that all users, regardless of their admin
-                                 ; status, are notified that the conference is muted when
-                                 ; the state is toggled.
-
-; participant_count        ; This action plays back the number of participants currently
-                           ; in a conference
-
-[sample_user_menu]
-type=menu
-*=playback_and_continue(conf-usermenu)
-*1=toggle_mute
-1=toggle_mute
-*4=decrease_listening_volume
-4=decrease_listening_volume
-*6=increase_listening_volume
-6=increase_listening_volume
-*7=decrease_talking_volume
-7=decrease_talking_volume
-*8=leave_conference
-8=leave_conference
-*9=increase_talking_volume
-9=increase_talking_volume
-
-[sample_admin_menu]
-type=menu
-*=playback_and_continue(conf-adminmenu)
-*1=toggle_mute
-1=toggle_mute
-*2=admin_toggle_conference_lock ; only applied to admin users
-2=admin_toggle_conference_lock  ; only applied to admin users
-*3=admin_kick_last       ; only applied to admin users
-3=admin_kick_last        ; only applied to admin users
-*4=decrease_listening_volume
-4=decrease_listening_volume
-*6=increase_listening_volume
-6=increase_listening_volume
-*7=decrease_talking_volume
-7=decrease_talking_volume
-*8=no_op
-8=no_op
-*9=increase_talking_volume
-9=increase_talking_volume

version_1.8-reactNative/pabx/etc/asterisk/config_test.conf

@@ -1,46 +0,0 @@
-; 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-reactNative/pabx/etc/asterisk/console.conf

@@ -1,97 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/dbsep.conf

@@ -1,34 +0,0 @@
-#
-# 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-reactNative/pabx/etc/asterisk/dnsmgr.conf

@@ -1,5 +0,0 @@
-[general]
-;enable=yes		; 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-reactNative/pabx/etc/asterisk/dsp.conf

@@ -1,42 +0,0 @@
-[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-reactNative/pabx/etc/asterisk/dundi.conf

@@ -1,275 +0,0 @@
-;
-; 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.  (It 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
-; in seconds. Default is 3600.
-;
-;cachetime=3600
-;
-; This defines the max depth (hops) 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 on' and 'dundi store history off') 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 (IAX2, SIP, H323)
-;
-; 'dest' is the Dial application's channel technology resource 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
-; ${SECRET}: The current IAX2 rotating secret key to be used
-;
-; Further options may include:
-;
-; nounsolicited:  No unsolicited calls of any type permitted via this
-;                 route
-; nocomunsolicit: No commercial unsolicited calls permitted via
-;                 this route
-; residential:    This number is known to be a residence
-; commercial:     This number is known to be a business
-; mobile:         This number is known to be a mobile phone
-; nocomunsolicit: No commercial unsolicited calls permitted via
-;                 this route
-; nopartial:      Do not search for partial matches
-;
-; There *must* exist an entry in mappings for DUNDi to respond
-; to any request, although it may be empty.
-;
-;empty_context =>
-;
-;e164 => dundi-e164-canonical,0,IAX2,dundi:${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/${NUMBER}
-
-;
-; Weights for mappings can be set a few different ways:
-;
-; 1) It can be set as a static number.
-;testmap1 => context1,222,IAX2,guest@peer1/${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 should 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 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 specifies the peer's entityid.  You can specify which
-; DUNDi contexts with which you want the trust to be established.
-;
-; inkey - What key they will be authenticating to us with
-;
-; outkey - What key we use to authenticate to them
-;
-; host - What their host is (DNS name, IP address, or dynamic)
-;
-; port - The port where their host is listening (default: 4520)
-;
-; ustothem - Explicitly specify the entityid we use with this peer.
-;
-; 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 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.
-;
-; qualify - Enable qualifying the peer to determine reachable status.
-;           Set to yes, no, or number of milliseconds for qualifying
-;           the peer's reachable status.
-;
-; register - Enable registering with the peer.  This presupposes that the
-;            peer's host option for us is dynamic.  (yes/no value)
-;
-; 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-reactNative/pabx/etc/asterisk/enum.conf

@@ -1,22 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/extconfig.conf

@@ -1,111 +0,0 @@
-;
-; 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-reactNative/pabx/etc/asterisk/extensions.ael

@@ -1,456 +0,0 @@
-//
-// 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-reactNative/pabx/etc/asterisk/extensions.conf

@@ -1,882 +0,0 @@
-; extensions.conf - the Asterisk dial plan
-;
-; Static extension configuration file, used by
-; the pbx_config module. This is where you configure all your
-; inbound and outbound calls in Asterisk.
-;
-; This configuration file is reloaded
-; - With the "dialplan reload" command in the CLI
-; - With the "reload" command (that reloads everything) in the CLI
-
-;
-; The "General" category is for certain variables.
-;
-[general]
-;
-; If static is set to no, or omitted, then the pbx_config will rewrite
-; this file when extensions are modified.  Remember that all comments
-; made in the file will be lost when that happens.
-;
-; XXX Not yet implemented XXX
-;
-static=yes
-;
-; if static=yes and writeprotect=no, you can save dialplan by
-; CLI command "dialplan save" too
-;
-writeprotect=no
-;
-; If autofallthrough is set, then if an extension runs out of
-; things to do, it will terminate the call with BUSY, CONGESTION
-; or HANGUP depending on Asterisk's best guess. This is the default.
-;
-; If autofallthrough is not set, then if an extension runs out of
-; things to do, Asterisk will wait for a new extension to be dialed
-; (this is the original behavior of Asterisk 1.0 and earlier).
-;
-;autofallthrough=no
-;
-;
-;
-; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
-; a Trie to find the best matching pattern is used. In dialplans
-; with more than about 20-40 extensions in a single context, this
-; new algorithm can provide a noticeable speedup.
-; With 50 extensions, the speedup is 1.32x
-; with 88 extensions, the speedup is 2.23x
-; with 138 extensions, the speedup is 3.44x
-; with 238 extensions, the speedup is 5.8x
-; with 438 extensions, the speedup is 10.4x
-; with 10,000 extensions, the speedup is 374x
-; Basically, the new algorithm provides a flat response
-; time, no matter the number of extensions.
-;
-; By default, the old pattern matcher is used.
-;
-; ****This is a new feature! *********************
-; The new pattern matcher is for the brave, the bold, and
-; the desperate. If you have large dialplans (more than about 50 extensions
-; in a context), and/or high call volume, you might consider setting
-; this value to "yes" !!
-; Please, if you try this out, and are forced to return to the
-; old pattern matcher, please report your reasons in a bug report
-; on https://issues.asterisk.org. We have made good progress in providing
-; something compatible with the old matcher; help us finish the job!
-;
-; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
-; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
-;
-;extenpatternmatchnew=no
-;
-; If clearglobalvars is set, global variables will be cleared
-; and reparsed on a dialplan reload, or Asterisk reload.
-;
-; If clearglobalvars is not set, then global variables will persist
-; through reloads, and even if deleted from the extensions.conf or
-; one of its included files, will remain set to the previous value.
-;
-; NOTE: A complication sets in, if you put your global variables into
-; the AEL file, instead of the extensions.conf file. With clearglobalvars
-; set, a "reload" will often leave the globals vars cleared, because it
-; is not unusual to have extensions.conf (which will have no globals)
-; load after the extensions.ael file (where the global vars are stored).
-; So, with "reload" in this particular situation, first the AEL file will
-; clear and then set all the global vars, then, later, when the extensions.conf
-; file is loaded, the global vars are all cleared, and then not set, because
-; they are not stored in the extensions.conf file.
-;
-clearglobalvars=no
-;
-; User context is where entries from users.conf are registered.  The
-; default value is 'default'
-;
-;userscontext=default
-;
-; You can include other config files, use the #include command
-; (without the ';'). Note that this is different from the "include" command
-; that includes contexts within other contexts. The #include command works
-; in all asterisk configuration files.
-;#include "filename.conf"
-;#include <filename.conf>
-;#include filename.conf
-;
-; You can execute a program or script that produces config files, and they
-; will be inserted where you insert the #exec command. The #exec command
-; works on all asterisk configuration files.  However, you will need to
-; activate them within asterisk.conf with the "execincludes" option.  They
-; are otherwise considered a security risk.
-;#exec /opt/bin/build-extra-contexts.sh
-;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
-;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
-;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
-;
-
-; The "Globals" category contains global variables that can be referenced
-; in the dialplan with the GLOBAL dialplan function:
-; ${GLOBAL(VARIABLE)}
-; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
-; Unix/Linux environmental variables can be reached with the ENV dialplan
-; function: ${ENV(VARIABLE)}
-;
-[globals]
-CONSOLE=Console/dsp				; Console interface for demo
-;CONSOLE=DAHDI/1
-;CONSOLE=Phone/phone0
-IAXINFO=guest					; IAXtel username/password
-;IAXINFO=myuser:mypass
-TRUNK=DAHDI/G2					; Trunk interface
-;
-; Note the 'G2' in the 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).
-;
-TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
-;TRUNK=IAX2/user:pass@provider
-
-;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
-                                                ; freenum calls (uses outbound-freenum
-                                                ; context)
-
-;
-; WARNING WARNING WARNING WARNING
-; If you load any other extension configuration engine, such as pbx_ael.so,
-; your global variables may be overridden by that file.  Please take care to
-; use only one location to set global variables, and you will likely save
-; yourself a ton of grief.
-; WARNING WARNING WARNING WARNING
-;
-; 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 parentheses after their name which can be used in goto situations.
-;
-; Contexts contain several lines, one for each step of each extension.  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.
-; Switches may also be included within a context.  The order of matching within
-; a context is always exact extensions, pattern match extensions, includes, and
-; switches.  Includes are always processed depth-first.  So for example, if you
-; would like a switch "A" to match before context "B", simply put switch "A" in
-; an included context "C", where "C" is included in your original context
-; before "B".
-;
-;[context]
-;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
-;
-; Timing list for includes is
-;
-;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
-;
-; Note that ranges may be specified to wrap around the ends.  Also, minutes are
-; fine-grained only down to the closest even minute.
-;
-;include => daytime,9:00-17:00,mon-fri,*,*
-;include => weekend,*,sat-sun,*,*
-;include => weeknights,17:02-8:58,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.  Please note that ignorepat
-; only works with channels which receive dialtone from the PBX, such as DAHDI,
-; Phone, and VPB.  Other channels, such as SIP and MGCP, which generate their
-; own dialtone and converse with the PBX only after a number is complete, are
-; generally unaffected by ignorepat (unless DISA or another method is used to
-; generate a dialtone after answering the channel).
-;
-
-;
-; Sample entries for extensions.conf
-;
-;
-[dundi-e164-canonical]
-;include => stdexten
-;
-; List canonical entries here
-;
-;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
-;exten => 12564286000,n,Goto(default,s,1)	; exited Voicemail
-;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
-
-[dundi-e164-customers]
-;
-; If you are an ITSP or Reseller, list your customers here.
-;
-;exten => _12564286000,1,Dial(SIP/customer1)
-;exten => _12564286001,1,Dial(IAX2/customer2)
-
-[dundi-e164-via-pstn]
-;
-; If you are freely delivering calls to the PSTN, list them here
-;
-;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
-;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
-
-[dundi-e164-local]
-;
-; Context to put your dundi IAX2 or SIP user in for
-; full access
-;
-include => dundi-e164-canonical
-include => dundi-e164-customers
-include => dundi-e164-via-pstn
-
-[dundi-e164-switch]
-;
-; Just a wrapper for the switch
-;
-switch => DUNDi/e164
-
-[dundi-e164-lookup]
-;
-; Locally to lookup, try looking for a local E.164 solution
-; then try DUNDi if we don't have one.
-;
-include => dundi-e164-local
-include => dundi-e164-switch
-;
-; DUNDi can also be implemented as a Macro instead of using
-; the Local channel driver.
-;
-[macro-dundi-e164]
-;
-; ARG1 is the extension to Dial
-;
-; Extension "s" is not a wildcard extension that matches "anything".
-; In macros, it is the start extension. In most other cases,
-; you have to goto "s" to execute that extension.
-;
-; Note: In old versions of Asterisk the PBX in some cases defaulted to
-; extension "s" when a given extension was wrong (like in AMI originate).
-; This is no longer the case.
-;
-; For wildcard matches, see above - all pattern matches start with
-; an underscore.
-exten => s,1,Goto(${ARG1},1)
-include => dundi-e164-lookup
-
-;
-; Here are the entries you need to participate in the IAXTEL
-; call routing system.  Most IAXTEL numbers begin with 1-700, but
-; there are exceptions.  For more information, and to sign
-; up, please go to www.gnophone.com or www.iaxtel.com
-;
-[iaxtel700]
-exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
-
-;
-; The SWITCH statement permits a server to share the dialplan with
-; another server. Use with care: Reciprocal switch statements are not
-; allowed (e.g. both A -> B and B -> A), and the switched server needs
-; to be on-line or else dialing can be severly delayed.
-;
-[iaxprovider]
-;switch => IAX2/user:[key]@myserver/mycontext
-
-[trunkint]
-;
-; International long distance through trunk
-;
-exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
-exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
-
-[trunkld]
-;
-; Long distance context accessed through trunk
-;
-exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
-exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[trunklocal]
-;
-; Local seven-digit dialing accessed through trunk interface
-;
-exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[trunktollfree]
-;
-; Long distance context accessed through trunk interface
-;
-exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[international]
-;
-; Master context for international long distance
-;
-ignorepat => 9
-include => longdistance
-include => trunkint
-
-[longdistance]
-;
-; Master context for long distance
-;
-ignorepat => 9
-include => local
-include => trunkld
-
-[local]
-;
-; Master context for local, toll-free, and iaxtel calls only
-;
-ignorepat => 9
-include => default
-include => trunklocal
-include => iaxtel700
-include => trunktollfree
-include => iaxprovider
-
-;Include parkedcalls (or the context you define in features conf)
-;to enable call parking.
-include => parkedcalls
-;
-; You can use an alternative switch type as well, to resolve
-; extensions that are not known here, for example with remote
-; IAX switching you transparently get access to the remote
-; Asterisk PBX
-;
-; switch => IAX2/user:password@bigserver/local
-;
-; An "lswitch" is like a switch but is literal, in that
-; variable substitution is not performed at load time
-; but is passed to the switch directly (presumably to
-; be substituted in the switch routine itself)
-;
-; lswitch => Loopback/12${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@${CURSERVER}
-
-; The following two contexts are a template to enable the ability to dial
-; ISN numbers. For more information about what an ISN number is, please see
-; http://www.freenum.org.
-;
-; This is the dialing hook.  use:
-; include => outbound-freenum
-
-[outbound-freenum]
-; We'll add more digits as needed. The purpose is to dial things
-; like extension numbers at domains (ITAD number) so we're matching
-; on lengths of 1 through 6 prior to the separator (the asterisk [*])
-;
-exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-
-[outbound-freenum2]
-; This is the handler which performs the dialing logic. It is called
-; from the [outbound-freenum] context
-;
-exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
-same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})                                ; make sure the suffix is all digits as well
-same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
-                                                                        ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
-same => n,Set(TIMEOUT(absolute)=10800)
-same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})     ; perform our lookup with freenum.org
-same => n,GotoIf($["${isnresult}" != ""]?from)
-same => n,Set(DIALSTATUS=CONGESTION)
-same => n,Goto(fn-CONGESTION,1)
-same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
-same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)               ; check if we set the FREENUMDOMAIN global variable in [global]
-same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})                 ;    if we did set it, then we'll use it for our outbound dialing domain
-same => n(dial),Dial(SIP/${isnresult},40)
-same => n,Goto(fn-${DIALSTATUS},1)
-
-exten => fn-BUSY,1,Busy()
-
-exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
-same => n,Congestion()
-
-[macro-trunkdial]
-;
-; Standard trunk dial macro (hangs up on a dialstatus that should
-; terminate call)
-;   ${ARG1} - What to dial
-;
-exten => s,1,Dial(${ARG1})
-exten => s,n,Goto(s-${DIALSTATUS},1)
-exten => s-NOANSWER,1,Hangup()
-exten => s-BUSY,1,Hangup()
-exten => _s-.,1,NoOp
-
-[stdexten]
-;
-; Standard extension subroutine:
-;   ${EXTEN} - Extension
-;   ${ARG1} - Device(s) to ring
-;   ${ARG2} - Optional context in Voicemail
-;
-; Note that the current version will drop through to the next priority in the
-; case of their pressing '#'.  This gives more flexibility in what do to next:
-; you can prompt for a new extension, or drop the call, or send them to a
-; general delivery mailbox, or...
-;
-; The use of the LOCAL() function is purely for convenience.  Any variable
-; initially declared as LOCAL() will disappear when the innermost Gosub context
-; in which it was declared returns.  Note also that you can declare a LOCAL()
-; variable on top of an existing variable, and its value will revert to its
-; previous value (before being declared as LOCAL()) upon Return.
-;
-exten => _X.,50000(stdexten),NoOp(Start stdexten)
-exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
-exten => _X.,n,Set(LOCAL(dev)=${ARG1})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
-exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
-exten => _X.,n,Dial(${dev},20)				; Ring the interface, 20 seconds maximum
-exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-
-exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
-exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
-
-exten => stdexten-BUSY,1,VoiceMail(${mbx},b)		; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,Return()			; If they press #, return to start
-
-exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
-
-exten => a,1,VoiceMailMain(${mbx})			; If they press *, send the user into VoicemailMain
-exten => a,n,Return()
-
-[stdPrivacyexten]
-;
-; Standard extension subroutine:
-;   ${ARG1} - Extension
-;   ${ARG2} - Device(s) to ring
-;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
-;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
-;   ${ARG5} - Context in voicemail (if empty, then "default")
-;
-; See above note in stdexten about priority handling on exit.
-;
-exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
-exten => _X.,n,Set(LOCAL(ext)=${ARG1})
-exten => _X.,n,Set(LOCAL(dev)=${ARG2})
-exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
-exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
-
-exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
-exten => _X.,n,Dial(${dev},20,p)			; Ring the interface, 20 seconds maximum, call screening
-						; option (or use P for databased call _X.creening)
-exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-
-exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
-exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
-exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
-
-exten => stdexten-BUSY,1,VoiceMail(${mbx},b)		; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
-exten => stdexten-BUSY,n,Return()			; If they press #, return to start
-
-exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)	; Callee chose to send this call to a polite "Don't call again" script.
-
-exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)	; Callee chose to send this call to a telemarketer torture script.
-
-exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
-
-exten => a,1,VoiceMailMain(${mbx})		; If they press *, send the user into VoicemailMain
-exten => a,n,Return()
-
-[macro-page]
-;
-; Paging macro:
-;
-;       Check to see if SIP device is in use and DO NOT PAGE if they are
-;
-;   ${ARG1} - Device to page
-
-exten => s,1,ChanIsAvail(${ARG1},s)			; s is for ANY call
-exten => s,n,GotoIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
-exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
-exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
-exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
-exten => s,n,Dial(${ARG1})
-exten => s,n(fail),Hangup()
-
-[internal_acme]
-
-exten => _20XX,1,NoOp(${CALLERID(num)})
-exten => _20XX,n,NoOp(${CALLERID(name)})
-exten => _20XX,n,Dial(SIP/2${EXTEN},60,tr)
-exten => _20XX,n,Hangup
-
-[ramais_ws]
-
-exten => _X.,1,NoOp(chamada entre ramais ws)
-exten => _X.,n,Wait(1)
-exten => _X.,n,Dial(SIP/${EXTEN},120,tr)
-exten => _X.,n,Hangup()
-
-exten => _X.,n,Set(TRANSFER_DESTINATION=${EXTEN})
-exten => _X.,n,Read(DTMF_TO_TRANSFER,beep,3)
-exten => _X.,n,GotoIf($[${DTMF_TO_TRANSFER}) = "#"]?transferir:nao_transferir)
-
-exten => _X.,n(transferir),NoOp(Transferindo chamada para ${TRANSFER_DESTINATION})
-exten => _X.,n,Macro(dial,${TRANSFER_DESTINATION})
-exten => _X.,n,Hangup()
-
-exten => _X.,n(nao_transferir),NoOp(Nao foi pressionado o # no DTFM)
-
-[demo]
-include => stdexten
-;
-; We start with what to do when a call first comes in.
-;
-;exten => s,1,Wait(2)                    ; Wait a second, just for fun
-;        same => n,Set(FOLDERAUDIO=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):0:4}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):4:2}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):6:2}/)
-;        same => n,NoOp(${FOLDERAUDIO})
-;        same => n,MixMonitor(/home/admin/${FOLDERAUDIO}/${UNIQUEID}.wav,,/var/lib/asterisk/agi-bin/wav2mp3 ^{UNIQUEID}.wav /home/admin/^{FOLDERAUDIO}/)
-;        same => n,Answer()                      ; Answer the line
-;        same => n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
-;        same => n,Set(TIMEOUT(response)=10)     ; Set Response Timeout to 10 seconds
-;        same => n(restart),BackGround(demo-congrats)    ; Play a congratulatory message
-;        same => n(instruct),BackGround(demo-instruct)   ; Play some instructions
-;        same => n,WaitExten()           ; Wait for an extension to be dialed.
-
-exten => _10XX,1,NoOp(>>>> Chamando Ramal ...)
-exten => _10XX,n,Set(FOLDERAUDIO=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):0:4}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):4:2}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S):6:2}/)
-exten => _10XX,n,NoOp(${FOLDERAUDIO})
-exten => _10XX,n,MixMonitor(/home/debian/${FOLDERAUDIO}/${UNIQUEID}.wav,,/var/lib/asterisk/agi-bin/wav2mp3 ^{UNIQUEID}.wav /home/debian/^{FOLDERAUDIO}/)
-exten => _10XX,n,Dial(SIP/${EXTEN:1},60,Tr)
-exten => _10XX,n,Hangup
-
-exten => _20XX,1,NoOp($CALLERID(num))
-exten => _20XX,n,NoOp($CALLERID(name))
-exten => _20XX,n,Hangup
-
-;exten => 1000,1,Goto(demo,s,1)
-
-;
-; We also create an example user, 1234, who is on the console and has
-; voicemail, etc.
-;
-exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
-					; (but skip if channel is not up)
-exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
-exten => 1234,n,Goto(default,s,1)		; exited Voicemail
-
-exten => 1235,1,VoiceMail(1234,u)		; Right to voicemail
-
-exten => 1236,1,Dial(Console/dsp)		; Ring forever
-exten => 1236,n,VoiceMail(1234,b)		; Unless busy
-
-;
-; # for when they're done with the demo
-;
-exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
-exten => #,n,Hangup()			; Hang them up.
-
-;
-; A timeout and "invalid extension rule"
-;
-exten => t,1,Goto(#,1)			; If they take too long, give up
-exten => i,1,Playback(invalid)		; "That's not valid, try again"
-
-;
-; Create an extension, 500, for dialing the
-; Asterisk demo.
-;
-exten => 500,1,Playback(demo-abouttotry)	; Let them know what's going on
-exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
-exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
-exten => 500,n,Goto(s,6)		; Return to the start over message.
-
-;
-; Create an extension, 600, for evaluating echo latency.
-;
-exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
-exten => 600,n,Echo()			; Do the echo test
-exten => 600,n,Playback(demo-echodone)	; Let them know it's over
-exten => 600,n,Goto(s,6)		; Start over
-
-;
-;	You can use the Macro Page to intercom a individual user
-exten => 76245,1,Macro(page,SIP/Grandstream1)
-; or if your peernames are the same as extensions
-exten => _7XXX,1,Macro(page,SIP/${EXTEN})
-;
-;
-; System Wide Page at extension 7999
-;
-exten => 7999,1,Set(TIMEOUT(absolute)=60)
-exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
-
-; Give voicemail at extension 8500
-;
-exten => 8500,1,VoiceMailMain()
-exten => 8500,n,Goto(s,6)
-;
-; Here's what a phone entry would look like (IXJ for example)
-;
-;exten => 1265,1,Dial(Phone/phone0,15)
-;exten => 1265,n,Goto(s,5)
-
-;
-;	The page context calls up the page macro that sets variables needed for auto-answer
-;	It is in is own context to make calling it from the Page() application as simple as
-;	Local/{peername}@page
-;
-[page]
-exten => _X.,1,Macro(page,SIP/${EXTEN})
-
-;[mainmenu]
-;
-; Example "main menu" context with submenu
-;
-;exten => s,1,Answer
-;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
-;exten => s,n,WaitExten
-;exten => 1,1,Goto(submenu,s,1)
-;exten => 2,1,Hangup
-;include => default
-;
-;[submenu]
-;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
-;exten => s,n,Wait,2
-;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
-;exten => s,n,WaitExten
-;exten => 1,1,Goto(default,steve,1)
-;exten => 2,1,Goto(default,mark,2)
-
-[public]
-;
-; ATTENTION: If your Asterisk is connected to the internet and you do
-; not have allowguest=no in sip.conf, everybody out there may use your
-; public context without authentication.  In that case you want to
-; double check which services you offer to the world.
-;
-include => demo
-
-[default]
-;
-; By default we include the demo.  In a production system, you
-; probably don't want to have the demo there.
-;
-include => demo
-
-;
-; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
-; Note that you must have a [sipprovider] section in sip.conf
-;
-;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
-
-; Real extensions would go here. Generally you want real extensions to be
-; 4 or 5 digits long (although there is no such requirement) and start with a
-; single digit that is fairly large (like 6 or 7) so that you have plenty of
-; room to overlap extensions and menu options without conflict.  You can alias
-; them with names, too, and use global variables
-
-;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
-;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
-;exten => 6245,n(dial),Dial(${HINT},20,rtT)	; Use hint as listed
-;exten => 6245,n,VoiceMail(6245,u)		; Voicemail (unavailable)
-;exten => 6245,s+1,Hangup			; s+1, same as n
-;exten => 6245,dial+101,VoiceMail(6245,b)	; Voicemail (busy)
-;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
-;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
-;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
-;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
-;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}
-
-;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
-						; assuming ${MARK} is something like DAHDI/2
-;exten => 6275,n,Goto(default,s,1)		; exited Voicemail
-;exten => mark,1,Goto(6275,1)			; alias mark to 6275
-;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
-						; Ditto for wil
-;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
-;exten => wil,1,Goto(6236,1)
-
-;If you want to subscribe to the status of a parking space, this is
-;how you do it. Subscribe to extension 6600 in sip, and you will see
-;the status of the first parking lot with this extensions' help
-;exten => 6600,hint,park:701@parkedcalls
-;exten => 6600,1,noop
-;
-;To subscribe to the availability of a free member in the 'markq' queue.
-;Note: '_avail' is added to the QueueName
-;exten => 8501,hint,Queue:markq_avail
-;exten => 8501,1,Queue(markq)
-;
-; You can also monitor the status of a queue by providing a hint for a
-; particular queue name.
-;exten => 8502,hint,Queue:markq
-;exten => 8502,1,Queue(markq)
-;
-; Some other handy things are an extension for checking voicemail via
-; voicemailmain
-;
-;exten => 8500,1,VoicemailMain
-;exten => 8500,n,Hangup
-;
-; Or a conference room (you'll need to edit meetme.conf to enable this room)
-;
-;exten => 8600,1,Meetme(1234)
-;
-; Or playing an announcement to the called party, as soon it answers
-;
-;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
-;
-
-; example of a compartmentalized company called "acme"
-;
-; this is the context that your incoming IAX/SIP trunk dumps you in...
-;[acme-incoming]
-;exten => s,1,Wait(1)
-;exten => s,n,Answer()
-;exten => s,n(menu),Playback(acme/vm-brief-menu)
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
-;
-;include  => acme-extens
-;
-;exten => i,1,Playback(vm-invalid)
-;exten => i,n,Goto(s,exten)			; optionally, transfer to operator
-;
-;exten => t,1,Goto(s,goodbye)
-;
-; this is the context our internal SIP hardphones use (see sip.conf)
-;
-;[acme-internal]
-;exten => s,1,Answer()
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
-;
-;include => trunkint
-;include => trunkld
-;include => trunklocal
-;
-;include => acme-extens
-;
-; you can test what your system sounds like to outside callers by dialing this
-;exten => 777,1,DISA(no-password,acme-incoming)
-;
-; grouping of acme's extensions... never used directly, always included.
-;
-;[acme-extens]
-;include => stdexten
-;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
-;exten => 111,n,Goto(s,exten)
-;
-;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
-;exten => 112,n,Goto(s,end)
-;
-; end of acme example
-
-;
-; Time context: you can patch this in via the following.
-;
-; [acme-internal]
-; ...
-; exten => 777,1,Gosub(time)
-; exten => 777,n,Hangup()
-;
-; ...
-; include => time
-;
-; Note: if you're geographically spread out, you can have SIP extensions
-; specify their own local timezone in sip.conf as:
-;
-; [boi]
-; type=friend
-; context=acme-internal
-; callerid="Boise Ofc. <2083451111>"
-; ...
-; ; use system-wide default timezone of MST7MDT
-;
-; [lws]
-; type=friend
-; context=acme-internal
-; callerid="Lewiston Ofc. <2087431111>"
-; ...
-; setvar=timezone=PST8PDT
-;
-; "timezone" isn't a 'reserved' name in any way, and other places where
-; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
-; require modification as well.  Note that voicemail.conf already has
-; a mechanism for timezones.
-;
-
-[time]
-exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
-; the amount of delay is set for English; you may need to adjust this time
-; for other languages if there's no pause before the synchronizing beep.
-exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
-exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
-exten => _X.,n,SayPhonetic(z)
-; use the timezone associated with the extension (sip only), or system-wide
-; default if one hasn't been set.
-exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
-exten => _X.,n,Playback(spy-local)
-exten => _X.,n,WaitUntil(${FUTURETIME})
-exten => _X.,n,Playback(beep)
-exten => _X.,n,Return()
-
-;
-; ANI context: use in the same way as "time" above
-;
-
-[ani]
-exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
-exten => _X.,n,Playback(vm-from)
-exten => _X.,n,SayDigits(${CALLERID(ani)})
-exten => _X.,n,Wait(1.25)
-exten => _X.,n,SayDigits(${CALLERID(ani)})	; playback again in case of missed digit
-exten => _X.,n,Return()
-
-; For more information on applications, just type "core show applications" at your
-; frien

version_1.8-reactNative/pabx/etc/asterisk/extensions.lua

@@ -1,240 +0,0 @@
-
-
-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-reactNative/pabx/etc/asterisk/extensions_minivm.conf

@@ -1,158 +0,0 @@
-; 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))

version_1.8-reactNative/pabx/etc/asterisk/features.conf

@@ -1,122 +0,0 @@
-;
-; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
-;
-
-; Note: From Asterisk 12 - All parking lot configuration is now done in res_parking.conf
-
-[general]
-;transferdigittimeout => 3      ; Number of seconds to wait between digits when transferring a call
-                                ; (default is 3 seconds)
-;xfersound = beep               ; to indicate an attended transfer is complete
-;xferfailsound = beeperr        ; to indicate a failed transfer
-;pickupexten = *8               ; Configure the pickup extension. (default is *8)
-;pickupsound = beep             ; to indicate a successful pickup (default: no sound)
-;pickupfailsound = beeperr      ; to indicate that the pickup failed (default: no sound)
-;featuredigittimeout = 1000     ; Max time (ms) between digits for
-                                ; feature activation  (default is 1000 ms)
-;recordingfailsound = beeperr   ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
-                                ; to be applied to the call. (default: no sound)
-;atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
-;atxferdropcall = no            ; If someone does an attended transfer, then hangs up before the transfer
-                                ; target answers, then by default, the system will try to call back the
-                                ; person that did the transfer.  If this is set to "yes", the ringing
-                                ; transfer target is immediately transferred to the transferee.
-;atxferloopdelay = 10           ; Number of seconds to sleep between retries (if atxferdropcall = no)
-;atxfercallbackretries = 2      ; Number of times to attempt to send the call back to the transferer.
-                                ; By default, this is 2.
-;transferdialattempts = 3       ; Number of times that a transferer may attempt to dial an extension before
-                                ; being kicked back to the original call.
-;transferretrysound = "beep"    ; Sound to play when a transferer fails to dial a valid extension.
-;transferinvalidsound = "beeperr" ; Sound to play when a transferer fails to dial a valid extension and is out of retries.
-;atxferabort = *1               ; cancel the attended transfer
-;atxfercomplete = *2            ; complete the attended transfer, dropping out of the call
-;atxferthreeway = *3            ; complete the attended transfer, but stay in the call. This will turn the call into a multi-party bridge
-;atxferswap = *4                ; swap to the other party. Once an attended transfer has begun, this options may be used multiple times
-
-; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
-; They can not be used while the remote party is ringing or in progress. If you require this feature you can use
-; chan_local in combination with Answer to accomplish it.
-
-[featuremap]
-;blindxfer => #1                ; Blind transfer  (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
-;disconnect => *0               ; Disconnect  (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
-;automon => *1                  ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
-;atxfer => *2                   ; Attended transfer  -- Make sure to set the T and/or t option in the Dial() or Queue()  app call!
-;parkcall => #72                ; Park call (one step parking)  -- Make sure to set the K and/or k option in the Dial() app call!
-;automixmon => *3               ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!
-
-[applicationmap]
-; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
-; defined here.  The value of DYNAMIC_FEATURES should be the names of the features
-; to allow the channel to use separated by '#'.  For example:
-;
-;    Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
-;
-; (Note: The two leading underscores allow these feature settings to be set
-;  on the outbound channels, as well.  Otherwise, only the original channel
-;  will have access to these features.)
-;
-; The syntax for declaring a dynamic feature is any of the following:
-;
-;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
-;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
-;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]
-
-;
-;  FeatureName   -> This is the name of the feature used when setting the
-;                   DYNAMIC_FEATURES variable to enable usage of this feature.
-;  DTMF_sequence -> This is the key sequence used to activate this feature.
-;  ActivateOn    -> This is the channel of the call that the application will be executed
-;                   on. Valid values are "self" and "peer". "self" means run the
-;                   application on the same channel that activated the feature. "peer"
-;                   means run the application on the opposite channel from the one that
-;                   has activated the feature.
-;  ActivatedBy   -> ActivatedBy is no longer honored.  The feature is activated by which
-;                   channel DYNAMIC_FEATURES includes the feature is on.  Use a pre-dial
-;                   handler to set different values for DYNAMIC_FEATURES on the channels.
-;                   Historic values are: "caller", "callee", and "both".
-;  Application   -> This is the application to execute.
-;  AppArguments  -> These are the arguments to be passed into the application.  If you need
-;                   commas in your arguments, you should use either the second or third
-;                   syntax, above.
-;  MOH_Class     -> This is the music on hold class to play while the idle
-;                   channel waits for the feature to complete. If left blank,
-;                   no music will be played.
-;
-
-;
-; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
-;   applications. When applications are used in extensions.conf, they are executed
-;   by the PBX core. In this case, these applications are executed outside of the
-;   PBX core, so it does *not* make sense to use any application which has any
-;   concept of dialplan flow. Examples of this would be things like Goto,
-;   Background, WaitExten, and many more.  The exceptions to this are Gosub and
-;   Macro routines which must complete for the call to continue.
-;
-; Enabling these features means that the PBX needs to stay in the media flow and
-; media will not be re-directed if DTMF is sent in the media stream.
-;
-; Example Usage:
-;
-;testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and callee to play
-;                                            ;tt-monkeys to the opposite channel
-;
-; Set arbitrary channel variables, based upon CALLERID number (Note that the application
-; argument contains commas)
-;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
-;
-;pauseMonitor   => #1,self/callee,Pausemonitor     ;Allow the callee to pause monitoring
-;                                                  ;on their channel
-;unpauseMonitor => #3,self/callee,UnPauseMonitor   ;Allow the callee to unpause monitoring
-;                                                  ;on their channel
-
-; Dynamic Feature Groups:
-;   Dynamic feature groups are groupings of features defined in [applicationmap]
-;   that can have their own custom key mappings.  To give a channel access to a dynamic
-;   feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
-;
-; example:
-; [myGroupName]         ; defines the group named myGroupName
-; testfeature => #9     ; associates testfeature with the group and the keycode '#9'.
-; pauseMonitor =>       ; associates pauseMonitor with the group and uses the keycode specified
-;                       ; in the [applicationmap].

version_1.8-reactNative/pabx/etc/asterisk/festival.conf

@@ -1,34 +0,0 @@
-;
-; Festival Configuration
-;
-[general]
-;
-; Host which runs the festival server (default : localhost);
-;
-;host=localhost
-;
-; Port on host where the festival server runs (default : 1314)
-;
-;port=1314
-;
-; Use cache (yes, no - defaults to no)
-;
-;usecache=yes
-;
-; If usecache=yes, a directory to store waveform cache files.
-; The cache is never cleared (yet), so you must take care of cleaning it
-; yourself (just delete any or all files from the cache).
-; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
-; Defaults to /tmp/
-;
-;cachedir=/var/lib/asterisk/festivalcache/
-;
-; Festival command to send to the server.
-; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n
-; %s is replaced by the desired text to say. The command MUST end with a
-; (quit) directive, or the cache handling mechanism will hang. Do not
-; forget the \n at the end.
-;
-;festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
-;
-;

version_1.8-reactNative/pabx/etc/asterisk/followme.conf

@@ -1,107 +0,0 @@
-; Find-Me / Follow-Me Configuration File
-[general]
-;
-featuredigittimeout=>5000
-; The number of ms to wait for a digit input for the callee on whether to take the call or
-; not before we consider them "done" entering digits.
-;
-enable_callee_prompt=>true
-; Enable prompting a callee to either accept or reject the forwarded call.
-; If disabled, the optional prompting for caller name (option 'a') is
-; likewise disabled. Enabled by default.
-;
-takecall=>1
-; The global default keypress for the callee to take taking the current call. This can be
-; a single digit or multiple digits. Default is "1".
-; Note this string must not be a substring of declinecall.
-;
-declinecall=>2
-; The global default keypress for the callee to decline taking the current call. This can
-; be a single digit or multiple digits. Default is "2".
-; Note this string must not be a substring of takecall.
-;
-call_from_prompt=>followme/call-from
-; The global default for the 'Incoming call from' message.
-;
-norecording_prompt=>followme/no-recording
-; The global default for the 'You have an incoming call' message when the caller elects
-; not to leave their name or the option isn't set for them to do so.
-;
-options_prompt=>followme/options
-; The global default for the 'Press 1 to accept this call or press 2 to decline it' message.
-;
-pls_hold_prompt=>followme/pls-hold-while-try
-; The global default for 'Please hold while we try and connect your call' message.
-;
-status_prompt=>followme/status
-; The global default for 'The party you're calling isn't at their desk' message.
-;
-sorry_prompt=>followme/sorry
-; The global default for 'I'm sorry, but we were unable to locate your party' message.
-;
-connecting_prompt=>
-; The global default sound file name for 'Please say hello to the caller' message.
-; Setting to an empty string skips playing the prompt.  The default is no prompt
-; file name.
-
-[default]
-musicclass=>default
-; The moh class that should be used for the caller while they are waiting to be connected.
-context=>default
-; The context to dial the numbers from
-number=>01233456,25
-; The a follow-me number to call. The format is:
-; number=> <number to call[&2nd #[&...]]>[,<timeout value in seconds>[,<order in follow-me>]]
-; You can specify as many of these numbers as you like. They will be dialed in the
-; order that you specify them in the config file OR as specified with the order field
-; on the number prompt. As you can see from the example, forked dialing of multiple
-; numbers in the same step is supported with this application if you'd like to dial
-; multiple numbers in the same followme step.
-;
-; The timeout value is the amount of time allowed between the time the dialing step
-; starts and the callee answers.  The callee then has until the timeout of the last
-; step to make a choice on whether to take the call or not.  That being the case,
-; you may want to make the timeout on the last step longer to give enough time to
-; make the choice to accept or not.
-enable_callee_prompt=>true
-; Enable prompting the callee to accept the forwarded call. The default
-; is the global value.
-takecall=>1
-; The keypress for the callee to take taking the current call. This can be
-; a single digit or multiple digits. Default is the global default.
-; Note this string must not be a substring of declinecall.
-;
-declinecall=>2
-; The keypress for the callee to decline taking the current call. This can
-; be a single digit or multiple digits. Default is the global default.
-; Note this string must not be a substring of takecall.
-;
-call_from_prompt=>followme/call-from
-; The 'Incoming call from' message prompt. Default is the global default.
-;
-norecording_prompt=>followme/no-recording
-; The 'You have an incoming call' message prompt when the caller elects
-; not to leave their name or the option isn't set for them to do so. Default
-; is the global default.
-;
-options_prompt=>followme/options
-; The 'Press 1 to accept this call or press 2 to decline it' message prompt.
-; Default is the global default.
-;
-pls_hold_prompt=>followme/pls-hold-while-try
-; The 'Please hold while we try and connect your call' message prompt.
-; Default is the global default.
-;
-status_prompt=>followme/status
-; The 'The party you're calling isn't at their desk' message prompt.
-; Default is the global default.
-;
-sorry_prompt=>followme/sorry
-; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
-; is the global default.
-;
-connecting_prompt=>
-; The sound file name for 'Please say hello to the caller' message. Default is the
-; global default.
-;
-

version_1.8-reactNative/pabx/etc/asterisk/func_odbc.conf

@@ -1,112 +0,0 @@
-;
-; func_odbc.conf
-;
-[general]
-;
-; Asterisk uses separate connections for every database operation.
-; If single_db_connection is enabled then func_odbc will use a single
-; database connection per DSN.
-; This option exists for those who expect that a second func_odbc call
-; works on the same connection. That allows you to do a LAST_INSERT_ID()
-; in a second func_odbc call.
-; Note that you'll need additional dialplan locks for this behaviour to work.
-; There are better ways: using stored procedures/functions instead.
-; This option is enabled by default.
-;single_db_connection=yes
-;
-;
-; Each context is a separately defined function.  By convention, all
-; functions are entirely uppercase, so the defined contexts should also
-; be all-uppercase, but there is nothing that enforces this.  All functions
-; are case-sensitive, however.
-;
-; For substitution, you have ${ARG1}, ${ARG2} ... ${ARGn}
-; for the arguments to each SQL statement.
-;
-; In addition, for write statements, you have ${VAL1}, ${VAL2} ... ${VALn}
-; parsed, just like arguments, for the values.  In addition, if you want the
-; whole value, never mind the parsing, you can get that with ${VALUE}.
-;
-;
-; If you have data which may potentially contain single ticks, you may wish
-; to use the dialplan function SQL_ESC() to escape the data prior to its
-; inclusion in the SQL statement.
-;
-;
-; The following options are available in this configuration file:
-;
-; dsn          An alias for "writehandle."
-; readhandle   A comma-separated list of DSNs (from res_odbc.conf) to use when
-;              executing the readsql statement.  Each DSN is tried, in
-;              succession, until the statement succeeds.  You may specify up to
-;              5 DSNs per function class.  If not specified, it will default to
-;              the value of "writehandle" or "dsn," if specified.
-; writehandle  A comma-separated list of DSNs (from res_odbc.conf) to use when
-;              executing the writesql statement.  The same rules apply as to
-;              readhandle.
-; readsql      The statement to execute when reading from the function class.
-; writesql     The statement to execute when writing to the function class.
-; insertsql    The statement to execute when writing to the function class
-;              succeeds, but initially indicates that 0 rows were affected.
-; prefix       Normally, all function classes are prefixed with "ODBC" to keep
-;              them uniquely named.  You may choose to change this prefix, which
-;              may be useful to segregate a collection of certain function
-;              classes from others.
-; escapecommas This option may be used to turn off the default behavior of
-;              escaping commas which occur within a field.  If commas are
-;              escaped (the default behavior), then fields containing commas
-;              will be treated as a single value when assigning to ARRAY() or
-;              HASH().  If commas are not escaped, then values will be separated
-;              at the comma within fields.  Please note that turning this option
-;              off is incompatible with the functionality of HASH().
-; synopsis     Appears in the synopsis field for the command
-;              'core show function <function name>'
-; syntax       Appears in the syntax field for the command
-;              'core show function <function name>'
-; mode         This option may be set to 'multirow' to allow the function
-;              specified to return more than a single row.  However, this
-;              changes the way that func_odbc normally works.  Instead of the
-;              invocation of the function returning a row, it returns an opaque
-;              ID, which may be passed to ODBC_FETCH() to return each row in
-;              turn.  ODBC_FETCH_STATUS returns SUCCESS or FAILURE, to indicate
-;              whether any results were stored, and you should call ODBC_Finish
-;              on the ID to clean up any remaining results when you are done
-;              with the query.  Also, the variable ODBCROWS is set initially,
-;              which may be used in an iterative fashion to return each row in
-;              the result.
-;              Please note that multirow queries are isolated to the channel,
-;              and rows may not be fetched outside of the channel where the
-;              query was initially performed.  Additionally, as the results are
-;              associated with a channel, mode=multirow is incompatible with
-;              the global space.
-; rowlimit     Rowlimit limits the total number of rows which can be stored for
-;              that query.  For mode=multirow, otherwise, func_odbc will
-;              attempt to store all rows in the resultset, up to the maximum
-;              amount of memory.  In normal mode, rowlimit can be set to allow
-;              additional rows to be fetched, rather than just the first one.
-;              These additional rows can be returned by using the name of the
-;              function which was called to retrieve the first row as an
-;              argument to ODBC_FETCH().
-
-
-; ODBC_SQL - Allow an SQL statement to be built entirely in the dialplan
-[SQL]
-writehandle=mysql1
-readsql=${ARG1}
-
-; ODBC_ANTISOLICIT - A blacklist for known solicitors.
-[ANTISOLICIT]
-dsn=mysql1,mysql2   ; Use mysql1 as the primary handle, but fall back to mysql2
-                    ; if mysql1 is down.  Supports up to 5 comma-separated
-                    ; DSNs.  "dsn" may also be specified as "readhandle" and
-                    ; "writehandle", if it is important to separate reads and
-                    ; writes to different databases.
-readsql=SELECT COUNT(*) FROM known_solicitors WHERE callerid='${SQL_ESC(${ARG1})}'
-syntax=<callerid>
-synopsis=Check if a specified callerid is contained in the known solicitors database
-
-; ODBC_PRESENCE - Retrieve and update presence
-[PRESENCE]
-writehandle=mysql1
-readsql=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}'
-writesql=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE id='${SQL_ESC(${ARG1})}'

version_1.8-reactNative/pabx/etc/asterisk/hep.conf

@@ -1,32 +0,0 @@
-;
-; res_hep Module configuration for Asterisk
-;
-
-;
-; Note that this configuration file is consumed by res_hep, which is responsible
-; for the HEPv3 protocol manipulation and managing the connection to the Homer
-; capture server. Additional modules provide specific messages to be sent to
-; the Homer server:
-;  - res_hep_pjsip: Send SIP messages transmitted/received by the PJSIP stack
-;  - res_hep_rtcp: Send RTCP information (all channels)
-;
-
-; All settings are currently set in the general section.
-[general]
-enabled = no                       ; Enable/disable forwarding of packets to a
-                                   ; HEP server. Default is "yes".
-capture_address = 192.168.1.1:9061 ; The address of the HEP capture server.
-capture_password = foo             ; If specified, the authorization passsword
-                                   ; for the HEP server. If not specified, no
-                                   ; authorization password will be sent.
-capture_id = 1234                  ; A unique integer identifier for this
-                                   ; server. This ID will be embedded sent
-                                   ; with each packet from this server.
-uuid_type = call-id                ; Specify the preferred source for the Homer
-                                   ; correlation UUID. Valid options are:
-                                   ; - 'call-id' for the PJSIP or chan_sip SIP
-                                   ;             Call-ID
-                                   ; - 'channel' for the Asterisk channel name
-                                   ; Note: If 'call-id' is specified but the
-                                   ; channel is not PJSIP or chan_sip then the
-                                   ; Asterisk channel name will be used instead.

version_1.8-reactNative/pabx/etc/asterisk/http.conf

@@ -1,133 +0,0 @@
-;
-; Asterisk Built-in mini-HTTP server
-;
-;
-; Note about Asterisk documentation:
-;   If Asterisk was installed from a tarball, then the HTML documentation should
-;   be installed in the static-http/docs directory which is
-;   (/var/lib/asterisk/static-http/docs) on linux by default.  If the Asterisk
-;   HTTP server is enabled in this file by setting the "enabled", "bindaddr",
-;   and "bindport" options, then you should be able to view the documentation
-;   remotely by browsing to:
-;       http://<server_ip>:<bindport>/static/docs/index.html
-;
-[general]
-;
-; The name of the server, advertised in both the Server field in HTTP
-; response message headers, as well as the <address /> element in certain HTTP
-; response message bodies. If not furnished here, "Asterisk/{version}" will be
-; used as a default value for the Server header field and the <address />
-; element. Setting this property to a blank value will result in the omission
-; of the Server header field from HTTP response message headers and the
-; <address /> element from HTTP response message bodies.
-;
-servername=Asterisk
-;
-; Whether HTTP/HTTPS interface is enabled or not.  Default is no.
-; This also affects manager/rawman/mxml access (see manager.conf)
-;
-enabled=yes
-;
-; Address to bind to, both for HTTP and HTTPS. You MUST specify
-; a bindaddr in order for the HTTP server to run. There is no
-; default value.
-;
-bindaddr=0.0.0.0
-;
-; Port to bind to for HTTP sessions (default is 8088)
-;
-bindport=8088
-;
-; Prefix allows you to specify a prefix for all requests
-; to the server.  The default is blank.  If uncommented
-; all requests must begin with /asterisk
-;
-;prefix=asterisk
-;
-; sessionlimit specifies the maximum number of httpsessions that will be
-; allowed to exist at any given time. (default: 100)
-;
-;sessionlimit=100
-;
-; session_inactivity specifies the number of milliseconds to wait for
-; more data over the HTTP connection before closing it.
-;
-; Default: 30000
-;session_inactivity=30000
-;
-; session_keep_alive specifies the number of milliseconds to wait for
-; the next HTTP request over a persistent connection.
-;
-; Set to 0 to disable persistent HTTP connections.
-; Default: 15000
-;session_keep_alive=15000
-;
-; Whether Asterisk should serve static content from static-http
-; Default wis no.
-;
-enable_static=yes
-;
-; Whether Asterisk should serve a status page showing the running
-; configuration of this built-in HTTP server.
-; Default is yes.
-;
-;enable_status=no
-;
-; Redirect one URI to another.  This is how you would set a
-; default page.
-;   Syntax: redirect=<from here> <to there>
-; For example, if you are using the Asterisk-gui,
-; it is convenient to enable the following redirect:
-;
-redirect = / /static/config/index.html
-;
-; HTTPS support. In addition to enabled=yes, you need to
-; explicitly enable tls, define the port to use,
-; and have a certificate somewhere.
-tlsenable=no          ; enable tls - default no.
-tlsbindaddr=0.0.0.0:8089    ; address and port to bind to - default is bindaddr and port 8089.
-
-tlscertfile=/etc/asterisk/keys/asterisk.pem
-tlsprivatekey=/etc/asterisk/keys/asterisk.pem
-
-; If no path is given for tlscertfile or tlsprivatekey, default is to look in current
-; directory. If no tlsprivatekey is given, default is to search tlscertfile for private key.
-;
-; To produce a certificate you can e.g. use openssl. This places both the cert and
-; private in same .pem file.
-; openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
-;
-; tlscipher=                             ; The list of allowed ciphers
-;                                        ; if none are specified the following cipher
-;                                        ; list will be used instead:
-; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
-; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
-; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
-; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
-; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
-; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
-; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
-; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
-; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
-;
-; tlsdisablev1=yes                ; Disable TLSv1 support - if not set this defaults to "yes"
-; tlsdisablev11=yes               ; Disable TLSv1.1 support - if not set this defaults to "no"
-; tlsdisablev12=yes               ; Disable TLSv1.2 support - if not set this defaults to "no"
-;
-; tlsservercipherorder=yes        ; Use the server preference order instead of the client order
-;                                 ; Defaults to "yes"
-;
-; The post_mappings section maps URLs to real paths on the filesystem.  If a
-; POST is done from within an authenticated manager session to one of the
-; configured POST mappings, then any files in the POST will be placed in the
-; configured directory.
-;
-;[post_mappings]
-;
-; NOTE: You need a valid HTTP AMI mansession_id cookie with the manager
-; config permission to POST files.
-;
-; In this example, if the prefix option is set to "asterisk", then using the
-; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
-;uploads = /var/lib/asterisk/uploads/
-;

version_1.8-reactNative/pabx/etc/asterisk/iax.conf

@@ -1,665 +0,0 @@
-;
-; Inter-Asterisk eXchange v2 (IAX2) Channel Driver configuration
-;
-; This configuration is read when the chan_iax2.so module is loaded, and is
-; re-read when the module is reloaded, such as when invoking the CLI command:
-;
-;     *CLI> iax2 reload
-;
-
-; General settings, like port number to bind to, and an option address (the
-; default is to bind to all local addresses).
-
-[general]
-
-; Listener Addresses
-;
-; Use the 'bindaddr' and 'bindport' options to specify on which address and port
-; the IAX2 channel driver will listen for incoming requests.
-;
-;
-
-;bindport=4569           ; The default port to listen on
-                         ; NOTE: bindport must be specified BEFORE bindaddr or
-                         ; may be specified on a specific bindaddr if followed by
-                         ; colon and port (e.g. bindaddr=192.168.0.1:4569) or for
-                         ; IPv6 the address needs to be in brackets then colon
-                         ; and port (e.g. bindaddr=[2001:db8::1]:4569).
-
-;bindaddr=192.168.0.1    ; You can specify 'bindaddr' more than once to bind to
-                         ; multiple addresses, but the first will be the
-                         ; default. IPv6 addresses are accepted.
-
-;
-; Set 'iaxcompat' to yes if you plan to use layered switches or some other
-; scenario which may cause some delay when doing a lookup in the dialplan. It
-; incurs a small performance hit to enable it. This option causes Asterisk to
-; spawn a separate thread when it receives an IAX2 DPREQ (Dialplan Request)
-; instead of blocking while it waits for a response.
-;
-; Accepted values: yes, no
-; Default value:   no
-;
-;iaxcompat=yes
-;
-
-;
-; Disable UDP checksums (if nochecksums is set, then no checkums will
-; be calculated/checked on systems supporting this feature)
-;
-; Accepted values: yes, no
-; Default value:   no
-;
-;nochecksums=yes
-;
-
-;
-; For increased security against brute force password attacks enable
-; 'delayreject' which will delay the sending of authentication reject for REGREQ
-; or AUTHREP if there is a password.
-;
-; Accepted values: yes, no
-; Default value:   no
-;
-;delayreject=yes
-;
-
-;
-; You may specify a global default AMA flag for iaxtel calls.  These flags are
-; used in the generation of call detail records.
-;
-; Accepted values: default, omit, billing, documentation
-; Default value:   default
-;
-;amaflags=billing
-;
-
-;
-; ADSI (Analog Display Services Interface) can be enabled if you have (or may
-; have) ADSI compatible CPE equipment.
-;
-; Accepted values: yes, no
-; Default value:   no
-;
-;adsi=yes
-;
-
-;
-; Whether or not to perform an SRV lookup on outbound calls.
-;
-; Accepted values: yes, no
-; Default value:   no
-;
-;srvlookup=yes
-;
-
-;
-; You may specify a default account for Call Detail Records (CDRs) in addition to
-; specifying on a per-user basis.
-;
-; Accepted values: Any string value up to 19 characters in length
-; Default value:   <empty>
-;
-;accountcode=lss0101
-;
-
-;
-; You may specify a global default language for users.  This can be specified
-; also on a per-user basis.  If omitted, will fallback to English (en).
-;
-; Accepted values: A language tag such as 'en' or 'es'
-; Default value:   en
-;
-;language=en
-;
-
-;
-; This option specifies a preference for which music-on-hold class this channel
-; should listen to when put on hold if the music class has not been 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 music class.
-;
-; If this option is set to "passthrough", then the hold message will always be
-; passed through as signalling instead of generating hold music locally.
-;
-; This option may be specified globally, or on a per-user or per-peer basis.
-;
-; Accepted values: passthrough, or any music-on-hold class name
-; Default value:   <empty>
-;
-;mohinterpret=default
-;
-
-;
-; The 'mohsuggest' option specifies which music on hold class to suggest to the
-; peer channel when this channel places the peer on hold. It may be specified
-; globally or on a per-user or per-peer basis.
-;
-;mohsuggest=default
-;
-
-;
-; Specify bandwidth of low, medium, or high to control which codecs are used
-; in general.
-;
-bandwidth=low
-;
-
-;
-; You can also fine tune codecs here using "allow" and "disallow" clauses with
-; specific codecs.  Use "all" to represent all formats.
-;
-;allow=all
-;disallow=g723.1
-disallow=lpc10
-;allow=gsm
-;
-
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-; Jitter Buffer
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-
-;
-; You can adjust several parameters relating to the jitter buffer.  The jitter
-; buffer's function is to compensate for varying network delay.
-;
-; All of the jitter buffer settings are in milliseconds.  The jitter buffer
-; works for INCOMING audio only - the outbound audio will be dejittered by the
-; jitter buffer at the other end.
-;
-; jitterbuffer=yes|no: global default as to whether you want
-; the jitter buffer at all.
-;
-; maxjitterbuffer: a maximum size for the jitter buffer.
-; Setting a reasonable maximum here will prevent the call delay
-; from rising to silly values in extreme situations; you'll hear
-; SOMETHING, even though it will be jittery.
-;
-; resyncthreshold: when the jitterbuffer notices a significant change in delay
-; that continues over a few frames, it will resync, assuming that the change in
-; delay was caused by a timestamping mix-up. The threshold for noticing a
-; change in delay is measured as twice the measured jitter plus this resync
-; threshold.
-; Resyncing can be disabled by setting this parameter to -1.
-;
-; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
-; should return in a row. Since some clients do not send CNG/DTX frames to
-; indicate silence, the jitterbuffer will assume silence has begun after
-; returning this many interpolations. This prevents interpolating throughout
-; a long silence.
-;
-; jittertargetextra: 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.
-;
-
-jitterbuffer=no
-;maxjitterbuffer=1000
-;maxjitterinterps=10
-;resyncthreshold=1000
-;jittertargetextra=40
-
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-; IAX2 Encryption
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-
-;
-; Enable IAX2 encryption.  The default is no.
-;
-;encryption=yes
-;
-
-;
-; Force encryption insures no connection is established unless both sides
-; support encryption.  By turning this option on, encryption is automatically
-; turned on as well.  The default is no.
-;
-;forceencryption=yes
-;
-
-; This option defines the maximum payload in bytes an IAX2 trunk can support at
-; a given time.  The best way to explain this is to provide an example.  If the
-; maximum number of calls to be supported is 800, and each call transmits 20ms
-; frames of audio using ulaw:
-;
-;     (8000hz / 1000ms) * 20ms * 1 byte per sample = 160 bytes per frame
-;
-; The maximum load in bytes is:
-;
-;     (160 bytes per frame) * (800 calls) = 128000 bytes
-;
-; Once this limit is reached, calls may be dropped or begin to lose audio.
-; Depending on the codec in use and number of channels to be supported this value
-; may need to be raised, but in most cases the default value is large enough.
-;
-; trunkmaxsize = 128000 ; defaults to 128000 bytes, which supports up to 800
-                        ; calls of ulaw at 20ms a frame.
-
-; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice
-; quality when allowing the Linux system to handle fragmentation of UDP packets.
-; Depending on the size of each payload, allowing the OS to handle fragmentation
-; may not be very efficient. This setting sets the maximum transmission unit for
-; IAX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload
-; is over 1240 bytes for every 20ms it will be broken into multiple 1240 byte
-; messages.  Zero disables this functionality and let's the OS handle
-; fragmentation.
-;
-; trunkmtu = 1240    ; trunk data will be sent in 1240 byte messages.
-
-; trunkfreq sets how frequently trunk messages are sent in milliseconds. This
-; value is 20ms by default, which means the trunk will send all the data queued
-; to it in the past 20ms.  By increasing the time between sending trunk messages,
-; the trunk's payload size will increase as well.  Note, depending on the size
-; set by trunkmtu, messages may be sent more often than specified.  For example
-; if a trunk's message size grows to the trunkmtu size before 20ms is reached
-; that message will be sent immediately.  Acceptable values are between 10ms and
-; 1000ms.
-;
-; trunkfreq=20    ; How frequently to send trunk msgs (in ms). This is 20ms by
-                  ; default.
-
-; Should we send timestamps for the individual sub-frames within trunk frames?
-; There is a small bandwidth use for these (less than 1kbps/call), but they
-; ensure that frame timestamps get sent end-to-end properly.  If both ends of
-; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
-; length for your codecs, you can probably suppress these.  The receiver must
-; also support this feature, although they do not also need to have it enabled.
-;
-; trunktimestamps=yes
-
-; Minimum and maximum amounts of time that IAX2 peers can request as a
-; registration expiration interval (in seconds).
-; minregexpire = 60
-; maxregexpire = 60
-
-; IAX2 helper threads
-
-; Establishes the number of iax helper threads to handle I/O.
-; iaxthreadcount = 10
-
-; Establishes the number of extra dynamic threads that may be spawned to handle I/O
-; iaxmaxthreadcount = 100
-
-;
-; We can register with another IAX2 server to let him know where we are
-; in case we have a dynamic IP address for example
-;
-; Register with tormenta using username marko and password secretpass
-;
-;register => marko:secretpass@tormenta.linux-support.net
-;
-; Register joe at remote host with no password
-;
-;register => joe@remotehost:5656
-;
-; Register marko at tormenta.linux-support.net using RSA key "torkey"
-;
-;register => marko:[torkey]@tormenta.linux-support.net
-;
-; Sample Registration for iaxtel
-;
-; Visit http://www.iaxtel.com to register with iaxtel.  Replace "user"
-; and "pass" with your username and password for iaxtel.  Incoming
-; calls arrive at the "s" extension of "default" context.
-;
-;register => user:pass@iaxtel.com
-;
-; Sample Registration for IAX2 + FWD
-;
-; To register using IAX2 with FWD, it must be enabled by visiting the URL
-; http://www.fwdnet.net/index.php?section_id=112
-;
-; Note that you need an extension in you default context which matches
-; your free world dialup number.  Please replace "FWDNumber" with your
-; FWD number and "passwd" with your password.
-;
-;register => FWDNumber:passwd@iax.fwdnet.net
-;
-; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
-; perceived external network address has changed.  When the stun_monitor is installed and
-; configured, chan_iax will renew all outbound registrations when the monitor detects any sort
-; of network change has occurred. By default this option is enabled, but only takes effect once
-; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
-; generate all outbound registrations on a network change, use the option below to disable
-; this feature.
-;
-; subscribe_network_change_event = yes ; on by default
-;
-; You can enable authentication debugging to increase the amount of
-; debugging traffic.
-;
-;authdebug = yes
-;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
-;tos=ef
-;cos=5
-;
-; If regcontext is specified, Asterisk will dynamically create and destroy
-; a NoOp priority 1 extension for a given peer who registers or unregisters
-; with us.  The actual extension is the 'regexten' parameter of the registering
-; peer or its name if 'regexten' is not provided.  More than one regexten
-; may be supplied if they are separated by '&'.  Patterns may be used in
-; regexten.
-;
-;regcontext=iaxregistrations
-;
-; If we don't get ACK to our NEW 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
-;
-; codecpriority controls the codec negotiation of an inbound IAX2 call.
-; This option is inherited to all user entities.  It can also be defined
-; in each user entity separately which will override the setting in general.
-;
-; The valid values are:
-;
-; caller   - Consider the callers preferred order ahead of the host's.
-; host     - Consider the host's preferred order ahead of the caller's.
-; disabled - Disable the consideration of codec preference altogether.
-;            (this is the original behaviour before preferences were added)
-; reqonly  - Same as disabled, only do not consider capabilities if
-;            the requested format is not available the call will only
-;            be accepted if the requested format is available.
-;
-; The default value is 'host'
-;
-;codecpriority=host
-;
-; allowfwdownload controls whether this host will serve out firmware to
-; IAX2 clients which request it.  This has only been used for the IAXy,
-; and it has been recently proven that this firmware distribution method
-; can be used as a source of traffic amplification attacks.  Also, the
-; IAXy firmware has not been updated for at least 18 months, so unless
-; you are provisioning IAXys in a secure network, we recommend that you
-; leave this option to the default, off.
-;
-;allowfwdownload=yes
-
-;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
-                    ; just like friends added from the config file only on a
-                    ; as-needed basis? (yes|no)
-
-;rtsavesysname=yes  ; Save systemname in realtime database at registration
-                    ; Default = no
-
-;rtupdate=yes       ; Send registry updates to database using realtime? (yes|no)
-                    ; If set to yes, when a IAX2 peer registers successfully,
-                    ; the IP address, the origination port, the registration period,
-                    ; and the username of the peer will be set to database via realtime.
-                    ; If not present, defaults to 'yes'.
-
-;rtautoclear=yes    ; Auto-Expire friends created on the fly on the same schedule
-                    ; as if it had just registered? (yes|no|<seconds>)
-                    ; If set to yes, when the registration expires, the friend will
-                    ; vanish from the configuration until requested again.
-                    ; If set to an integer, friends expire within this number of
-                    ; seconds instead of the registration interval.
-
-;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
-                       ; has expired based on its registration interval, used the stored
-                       ; address information regardless. (yes|no)
-
-;parkinglot=edvina     ; Default parkinglot for IAX2 peers and users
-                       ; This can also be configured per device
-                       ; Parkinglots are defined in features.conf
-
-;
-; The following two options are used to disable call token validation for the
-; purposes of interoperability with IAX2 endpoints that do not yet support it.
-;
-; Call token validation can be set as optional for a single IP address or IP
-; address range by using the 'calltokenoptional' option. 'calltokenoptional' is
-; only a global option.
-;
-;calltokenoptional=209.16.236.73/255.255.255.0
-;
-; By setting 'requirecalltoken=no', call token validation becomes optional for
-; that peer/user.  By setting 'requirecalltoken=auto', call token validation
-; is optional until a call token supporting peer registers successfully using
-; call token validation.  This is used as an indication that from now on, we
-; can require it from this peer.  So, requirecalltoken is internally set to yes.
-; requirecalltoken may only be used in peer/user/friend definitions,
-; not in the global scope.
-; By default, 'requirecalltoken=yes'.
-;
-;requirecalltoken=no
-;
-
-;
-; These options are used to limit the amount of call numbers allocated to a
-; single IP address.  Before changing any of these values, it is highly encouraged
-; to read the user guide associated with these options first.  In most cases, the
-; default values for these options are sufficient.
-;
-; The 'maxcallnumbers' option limits the amount of call numbers allowed for each
-; individual remote IP address.  Once an IP address reaches it's call number
-; limit, no more new connections are allowed until the previous ones close.  This
-; option can be used in a peer definition as well, but only takes effect for
-; the IP of a dynamic peer after it completes registration.
-;
-;maxcallnumbers=512
-;
-; The 'maxcallnumbers_nonvalidated' is used to set the combined number of call
-; numbers that can be allocated for connections where call token  validation
-; has been disabled.  Unlike the 'maxcallnumbers' option, this limit is not
-; separate for each individual IP address.  Any connection resulting in a
-; non-call token validated call number being allocated contributes to this
-; limit.  For use cases, see the call token user guide.  This option's
-; default value of 8192 should be sufficient in most cases.
-;
-;maxcallnumbers_nonvalidated=1024
-;
-; The [callnumberlimits] section allows custom call number limits to be set
-; for specific IP addresses and IP address ranges.  These limits take precedence
-; over the global 'maxcallnumbers' option, but may still be overridden by a
-; peer defined 'maxcallnumbers' entry.  Note that these limits take effect
-; for every individual address within the range, not the range as a whole.
-;
-;[callnumberlimits]
-;10.1.1.0/255.255.255.0 = 24
-;10.1.2.0/255.255.255.0 = 32
-;
-
-; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
-; in square brackets.  For example, the Caller*ID value 555.5555 becomes 5555555
-; when this option is enabled.  Disabling this option results in no modification
-; of the Caller*ID value, which is necessary when the Caller*ID represents something
-; that must be preserved.  This option can only be used in the [general] section.
-; By default this option is on.
-;
-;shrinkcallerid=yes     ; on by default
-
-; Guest sections for unauthenticated connection attempts.  Just specify an
-; empty secret, or provide no secret section.
-;
-[guest]
-type=user
-context=public
-callerid="Guest IAX User"
-
-;
-; Trust Caller*ID Coming from iaxtel.com
-;
-[iaxtel]
-type=user
-context=default
-auth=rsa
-inkeys=iaxtel
-
-;
-; Trust Caller*ID Coming from iax.fwdnet.net
-;
-[iaxfwd]
-type=user
-context=default
-auth=rsa
-inkeys=freeworlddialup
-
-;
-; Trust Caller*ID delivered over DUNDi/e164
-;
-;[dundi]
-;type=user
-;dbsecret=dundi/secret
-;context=dundi-e164-local
-
-;
-; Further user sections may be added, specifying a context and a secret used
-; for connections with that given authentication name.  Limited IP based
-; access control is allowed by use of "permit", "deny", and "acl" keywords.
-; Multiple rules are permitted. Multiple permitted contexts may be specified,
-; in which case the first will be the default.  You can also override
-; Caller*ID so that when you receive a call you set the Caller*ID to be what
-; you want instead of trusting what the remote user provides
-;
-; There are three authentication methods that are supported:  md5, plaintext,
-; and rsa.  The least secure is "plaintext", which sends passwords cleartext
-; across the net.  "md5" uses a challenge/response md5 sum arrangement, but
-; still requires both ends have plain text access to the secret.  "rsa" allows
-; unidirectional secret knowledge through public/private keys.  If "rsa"
-; authentication is used, "inkeys" is a list of acceptable public keys on the
-; local system that can be used to authenticate the remote peer, separated by
-; the ":" character.  "outkey" is a single, private key to use to authenticate
-; to the other side.  Public keys are named /var/lib/asterisk/keys/<name>.pub
-; while private keys are named /var/lib/asterisk/keys/<name>.key.  Private
-; keys should always be 3DES encrypted.
-;
-;
-; NOTE: All hostnames and IP addresses in this file are for example purposes
-;       only; you should not expect any of them to actually be available for
-;       your use.
-;
-;[markster]
-;type=user
-;context=default
-;context=local
-;auth=md5,plaintext,rsa
-;secret=markpasswd
-;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
-                                                ; cause the given audio file to
-                                                ; be played upon completion of
-                                                ; an attended transfer to the
-                                                ; target of the transfer.
-;dbsecret=mysecrets/place    ; Secrets can be stored in astdb, too
-;transfer=no                 ; Disable IAX2 native transfer
-;transfer=mediaonly          ; When doing IAX2 native transfers, transfer only
-                             ; the media stream
-;jitterbuffer=yes            ; Override the global setting and enable the jitter
-                             ; buffer for this user
-;maxauthreq=10               ; Set the maximum number of outstanding AUTHREQs
-                             ; waiting for replies. If this limit is reached,
-                             ; any further authentication will be blocked, until
-                             ; the pending requests expire or a reply is
-                             ; received.
-;callerid="Mark Spencer" <(256) 428-6275>
-;deny=0.0.0.0/0.0.0.0
-;accountcode=markster0101
-;permit=209.16.236.73/255.255.255.0
-;language=en                 ; Use english as default language
-;encryption=yes              ; Enable IAX2 encryption.  The default is no.
-;keyrotate=off               ; This is a compatibility option for older versions
-                             ; of IAX2 that do not support key rotation with
-                             ; encryption.  This option will disable the
-                             ; IAX_COMMAND_RTENC message.  The default is on.
-
-;
-; Peers may also be specified, with a secret and a remote hostname.
-;
-;[demo]
-;type=peer
-;username=asterisk
-;secret=supersecret
-;host=192.168.10.10
-;description=My IAX2 Peer            ; Description of this peer, as listed by
-                                     ; 'iax2 show peers'
-;sendani=no
-;host=asterisk.linux-support.net
-;port=5036
-;mask=255.255.255.255
-;qualify=yes                ; Make sure this peer is alive.
-;qualifysmoothing = yes     ; Use an average of the last two PONG results to
-                            ; reduce falsely detected LAGGED hosts.  The default
-                            ; is 'no.'
-;qualifyfreqok = 60000      ; How frequently to ping the peer when everything
-                            ; seems to be OK, in milliseconds.
-;qualifyfreqnotok = 10000   ; How frequently to ping the peer when it's either
-                            ; LAGGED or UNAVAILABLE, in milliseconds.
-;jitterbuffer=no            ; Turn off jitter buffer for this peer
-;
-;encryption=yes             ; Enable IAX2 encryption.  The default is no.
-;keyrotate=off              ; This is a compatibility option for older versions
-                            ; of IAX2 that do not support key rotation with
-                            ; encryption.  This option will disable the
-                            ; IAX_COMMAND_RTENC message.  The default is 'on.'
-
-; Peers can remotely register as well, so that they can be mobile.  Default
-; IPs can also optionally be given but are not required.  Caller*ID can be
-; suggested to the other side as well if it is for example a phone instead of
-; another PBX.
-;connectedline=yes ; Set if connected line and redirecting information updates
-;                  ; are passed between Asterisk servers for this peer.
-;                  ; yes - Sending and receiving updates are enabled.
-;                  ; send - Only send updates.
-;                  ; receive - Only process received updates.
-;                  ; no - Sending and receiving updates are disabled.
-;                  ; Default is "no".
-;                  ;
-;                  ; Note: Because of an incompatibility between Asterisk v1.4
-;                  ; and Asterisk v1.8 or later, this option must be set
-;                  ; to "no" toward the Asterisk v1.4 peer.  A symptom of the
-;                  ; incompatibility is the call gets disconnected unexpectedly.
-
-
-;[dynamichost]
-;host=dynamic
-;secret=mysecret
-; Note: app_voicemail mailboxes must be in the form of mailbox@context.
-;mailbox=1234		; Notify about mailbox 1234
-;inkeys=key1:key2
-;peercontext=local	; Default context to request for calls to peer
-;defaultip=216.207.245.34
-;callerid="Some Host" <(256) 428-6011>
-
-;[biggateway]
-;type=peer
-;host=192.168.0.1
-;description=Gateway to PSTN
-;context=*
-;secret=myscret
-;trunk=yes			; Use IAX2 trunking with this host
-;timezone=America/New_York	; Set a timezone for the date/time IE
-
-;
-; Friends are a shortcut for creating a user and a peer with the same values.
-;
-
-;[marko]
-;type=friend
-;host=dynamic
-;regexten=1234
-;secret=moofoo   ; Multiple secrets may be specified. For a "user", all
-;secret=foomoo   ; specified entries will be accepted as valid. For a "peer",
-;secret=shazbot  ; only the last specified secret will be used.
-;context=default
-;permit=0.0.0.0/0.0.0.0
-;acl=example_named_acl
-
-;
-; With immediate=yes, an IAX2 phone or a phone on an IAXy acts as a hot-line
-; which goes immediately to the s extension when picked up.  Useful for
-; elevator phones, manual service, or other similar applications.
-;
-;[manual]
-;type=friend
-;host=dynamic
-;immediate=yes  ; go immediately to s extension when picked up
-;secret=moofoo	; when immediate=yes is specified, secret is required
-;context=number-please ; we start at the s extension in this context
-;

version_1.8-reactNative/pabx/etc/asterisk/iaxprov.conf

@@ -1,80 +0,0 @@
-;
-; IAX2 Provisioning Information
-;
-; Contains provisioning information for templates and for specific service
-; entries.
-;
-; Templates provide a group of settings from which provisioning takes place.
-; A template may be based upon any template that has been specified before
-; it.  If the template that an entry is based on is not specified then it is
-; presumed to be 'default' (unless it is the first of course).
-;
-; Templates which begin with 'si-' are used for provisioning units with
-; specific service identifiers.  For example the entry "si-000364000126"
-; would be used when the device with the corresponding service identifier of
-; "000364000126" attempts to register or make a call.
-;
-[default]
-;
-; The port number the device should use to bind to.  The default is 4569.
-;
-;port=4569
-;
-; server is our PRIMARY server for registration and placing calls
-;
-;server=192.168.69.3
-;
-; altserver is the BACKUP server for registration and placing calls in the
-; event the primary server is unavailable.
-;
-;altserver=192.168.69.4
-;
-; port is the port number to use for IAX2 outbound.  The connections to the
-; server and altserver -- default is of course 4569.
-;serverport=4569
-;
-; language is the preferred language for the device
-;
-;language=en
-;
-; codec is the requested codec.  The iaxy supports ulaw and adpcm
-;
-codec=ulaw
-;
-; flags is a comma separated list of flags which the device should
-; use and may contain any of the following keywords:
-;
-;  "register"  - Register with server
-;  "secure"    - Do not accept calls / provisioning not originated by the server
-;  "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
-;  "debug"     - Output extra debugging to port 9999
-;
-;  Note that use can use += and -= to adjust parameters
-;
-flags=register,heartbeat
-;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
-;tos=ef
-;
-; Example iaxy provisioning
-;
-;[si-000364000126]
-;user=iaxy
-;pass=bitsy
-;flags += debug
-
-;[si-000364000127]
-;user=iaxy2
-;pass=bitsy2
-;template=si-000364000126
-;flags += debug
-
-;
-;[*]
-;
-;  If specified, the '*' provisioning is used for all devices which do not
-;  have another provisioning entry within the file.  If unspecified, no
-;  provisioning will take place for devices which have no entry.  DO NOT
-;  USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
-;
-;template=default

version_1.8-reactNative/pabx/etc/asterisk/indications.conf

@@ -1,747 +0,0 @@
-;
-; indications.conf
-;
-; Configuration file for location specific tone indications
-;
-
-;
-; NOTE:
-;    When adding countries to this file, please keep them in alphabetical
-;    order according to the 2-character country codes!
-;
-; The [general] category is for certain global variables.
-; All other categories are interpreted as location specific indications
-;
-
-[general]
-country=us		; default location
-
-
-; [example]
-; description = string
-;      The full name of your country, in English.
-; ringcadence = num[,num]*
-;      List of durations the physical bell rings.
-; dial = tonelist
-;      Set of tones to be played when one picks up the hook.
-; busy = tonelist
-;      Set of tones played when the receiving end is busy.
-; congestion = tonelist
-;      Set of tones played when there is some congestion (on the network?)
-; callwaiting = tonelist
-;      Set of tones played when there is a call waiting in the background.
-; dialrecall = tonelist
-;      Not well defined; many phone systems play a recall dial tone after hook
-;      flash.
-; record = tonelist
-;      Set of tones played when call recording is in progress.
-; info = tonelist
-;      Set of tones played with special information messages (e.g., "number is
-;      out of service")
-; 'name' = tonelist
-;      Every other variable will be available as a shortcut for the "PlayList" command
-;      but will not be used automatically by Asterisk.
-;
-;
-; The tonelist itself is defined by a comma-separated sequence of elements.
-; Each element consist of a frequency (f) with an optional duration (in ms)
-; attached to it (f/duration). The frequency component may be a mixture of two
-; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
-; The implicit modulation depth is fixed at 90%, though.
-; If the list element starts with a !, that element is NOT repeated,
-; therefore, only if all elements start with !, the tonelist is time-limited,
-; all others will repeat indefinitely.
-;
-; concisely:
-;   element = [!]freq[+|*freq2][/duration]
-;   tonelist = element[,element]*
-;
-
-[at]
-description = Austria
-ringcadence = 1000,5000
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-dial = 420
-busy = 420/400,0/400
-ring = 420/1000,0/5000
-congestion = 420/200,0/200
-callwaiting = 420/40,0/1960
-dialrecall = 420
-; RECORDTONE - not specified
-record = 1400/80,0/14920
-info = 950/330,1450/330,1850/330,0/1000
-stutter = 380+420
-
-[au]
-description = Australia
-; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
-; Normal Ring
-ringcadence = 400,200,400,2000
-; Distinctive Ring 1 - Forwarded Calls
-; 400,400,200,200,400,1400
-; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
-; 400,400,200,2000
-; Distinctive Ring 3 - Multiple Subscriber Number 1
-; 200,200,400,2200
-; Distinctive Ring 4 - Selective Ring 1 + Centrex
-; 400,2600
-; Distinctive Ring 5 - Selective Ring 3
-; 400,400,200,400,200,1400
-; Distinctive Ring 6 - Multiple Subscriber Number 2
-; 200,400,200,200,400,1600
-; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
-; 200,400,200,400,200,1600
-; Tones
-dial = 413+438
-busy = 425/375,0/375
-ring = 413+438/400,0/200,413+438/400,0/2000
-; XXX Congestion: Should reduce by 10 db every other cadence XXX
-congestion = 425/375,0/375,420/375,0/375
-callwaiting = 425/200,0/200,425/200,0/4400
-dialrecall = 413+438
-; Record tone used for Call Intrusion/Recording or Conference
-record = !425/1000,!0/15000,425/360,0/15000
-info = 425/2500,0/500
-; Other Australian Tones
-; The STD "pips" indicate the call is not an untimed local call
-std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
-; Facility confirmation tone (eg. Call Forward Activated)
-facility = 425
-; Message Waiting "stutter" dialtone
-stutter = 413+438/100,0/40
-; Ringtone for calls to Telstra mobiles
-ringmobile = 400+450/400,0/200,400+450/400,0/2000
-
-[bg]
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-description = Bulgaria
-ringcadence = 1000,4000
-;
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/150,0/150,425/150,0/4000
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-record = 1400/425,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425/1500,0/100
-
-[br]
-description = Brazil
-ringcadence = 1000,4000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/4000
-congestion = 425/250,0/250,425/750,0/250
-callwaiting = 425/50,0/1000
-; Dialrecall not used in Brazil standard (using UK standard)
-dialrecall = 350+440
-; Record tone is not used in Brazil, use busy tone
-record = 425/250,0/250
-; Info not used in Brazil standard (using UK standard)
-info = 950/330,1400/330,1800/330
-stutter = 350+440
-
-[be]
-description = Belgium
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,3000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/3000
-congestion = 425/167,0/167
-callwaiting = 1400/175,0/175,1400/175,0/3500
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = 900/330,1400/330,1800/330,0/1000
-stutter = 425/1000,0/250
-
-[ch]
-description = Switzerland
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/200,0/200,425/200,0/4000
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425+340/1100,0/1100
-
-[cl]
-description = Chile
-; According to specs from Telefonica CTC Chile
-ringcadence = 1000,3000
-dial = 400
-busy = 400/500,0/500
-ring = 400/1000,0/3000
-congestion = 400/200,0/200
-callwaiting = 400/250,0/8750
-dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-record = 1400/500,0/15000
-info = 950/333,1400/333,1800/333,0/1000
-stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-
-[cn]
-description = China
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 450
-busy = 450/350,0/350
-ring = 450/1000,0/4000
-congestion = 450/700,0/700
-callwaiting = 450/400,0/4000
-dialrecall = 450
-record = 950/400,0/10000
-info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
-; STUTTER - not specified
-stutter = 450+425
-
-[cz]
-description = Czech Republic
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425/330,0/330,425/660,0/660
-busy = 425/330,0/330
-ring = 425/1000,0/4000
-congestion = 425/165,0/165
-callwaiting = 425/330,0/9000
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
-; RECORDTONE - not specified
-record = 1400/500,0/14000
-info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
-; STUTTER - not specified
-stutter = 425/450,0/50
-
-[de]
-description = Germany
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/480,0/480
-ring = 425/1000,0/4000
-congestion = 425/240,0/240
-callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425+400
-
-[dk]
-description = Denmark
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-; STUTTER - not specified
-stutter = 425/450,0/50
-
-[ee]
-description = Estonia
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-; CALLWAIT not in accordance to ITU
-callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-; DIALRECALL - not specified
-dialrecall = 425/650,0/25
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO not in accordance to ITU
-info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-; STUTTER not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[es]
-description = Spain
-ringcadence = 1500,3000
-dial = 425
-busy = 425/200,0/200
-ring = 425/1500,0/3000
-congestion = 425/200,0/200,425/200,0/200,425/200,0/600
-callwaiting = 425/175,0/175,425/175,0/3500
-dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
-record = 1400/500,0/15000
-info = 950/330,0/1000
-dialout = 500
-; STUTTER not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-
-[fi]
-description = Finland
-ringcadence = 1000,4000
-dial = 425
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/8000
-dialrecall = 425/650,0/25
-record = 1400/500,0/15000
-info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-stutter = 425/650,0/25
-
-[fr]
-description = France
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1500,3500
-; Dialtone can also be 440+330
-dial = 440
-busy = 440/500,0/500
-ring = 440/1500,0/3500
-; CONGESTION - not specified
-congestion = 440/250,0/250
-callwait = 440/300,0/10000
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330
-stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
-
-[gr]
-description = Greece
-ringcadence = 1000,4000
-dial = 425/200,0/300,425/700,0/800
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/8000
-dialrecall = 425/650,0/25
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 425/650,0/25
-
-[hu]
-description = Hungary
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1250,3750
-dial = 425
-busy = 425/300,0/300
-ring = 425/1250,0/3750
-congestion = 425/300,0/300
-callwaiting = 425/40,0/1960
-dialrecall = 425+450
-; RECORDTONE - not specified
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 350+375+400
-
-[id]
-description = Indonesia
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/150,0/150,425/150,0/10000
-info = !950/330,!1400/330,!1800/330,0/1000
-
-[il]
-description = Israel
-ringcadence = 1000,3000
-dial = 414
-busy = 414/500,0/500
-ring = 414/1000,0/3000
-congestion = 414/250,0/250
-callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000
-dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414
-record = 1400/500,0/15000
-info = 1000/330,1400/330,1800/330,0/1000
-stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414
-
-
-[in]
-description = India
-ringcadence = 400,200,400,2000
-dial = 400*25
-busy = 400/750,0/750
-ring = 400*25/400,0/200,400*25/400,0/2000
-congestion = 400/250,0/250
-callwaiting = 400/200,0/100,400/200,0/7500
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,400*25
-
-[it]
-description = Italy
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425/200,0/200,425/600,0/1000
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
-dialrecall = 470/400,425/400
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 470/400,425/400
-
-[lt]
-description = Lithuania
-ringcadence = 1000,4000
-dial = 425
-busy = 425/350,0/350
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/4000
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[jp]
-description = Japan
-ringcadence = 1000,2000
-dial = 400
-busy = 400/500,0/500
-ring = 400+15/1000,0/2000
-congestion = 400/500,0/500
-callwaiting = 400+16/500,0/8000
-dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-
-[mx]
-description = Mexico
-ringcadence = 2000,4000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/200,0/600,425/200,0/10000
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
-
-[my]
-description = Malaysia
-ringcadence = 2000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/400,0/200,425/400,0/2000
-congestion = 425/500,0/500
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[nl]
-description = Netherlands
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-; Most of these 425's can also be 450's
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/500,0/9500
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425/500,0/50
-
-[no]
-description = Norway
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/200,0/600,425/200,0/10000
-dialrecall = 470/400,425/400
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 470/400,425/400
-
-[nz]
-description = New Zealand
-; Reference = http://www.telepermit.co.nz/TNA102.pdf
-ringcadence = 400,200,400,2000
-dial = 400
-busy = 400/500,0/500
-ring = 400+450/400,0/200,400+450/400,0/2000
-congestion = 400/250,0/250
-callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
-dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-record = 1400/425,0/15000
-info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
-stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400
-
-[ph]
-
-; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-
-description = Philippines
-ringcadence = 1000,4000
-dial = 425
-busy = 480+620/500,0/500
-ring = 425+480/1000,0/4000
-congestion = 480+620/250,0/250
-callwaiting = 440/300,0/10000
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO - not specified
-info = !950/330,!1400/330,!1800/330,0
-; STUTTER - not specified
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
-
-
-[pl]
-description = Poland
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/500,0/500
-callwaiting = 425/150,0/150,425/150,0/4000
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; 950/1400/1800 3x0.33 on 1.0 off  repeated 3 times
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[pt]
-description = Portugal
-ringcadence = 1000,5000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/5000
-congestion = 425/200,0/200
-callwaiting = 440/300,0/10000
-dialrecall = 425/1000,0/200
-record = 1400/500,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[ru]
-; References:
-;	http://www.minsvyaz.ru/site.shtml?id=1806
-;	http://www.aboutphone.info/lib/gost/45-223-2001.html
-description = Russian Federation / ex Soviet Union
-ringcadence = 1000,4000
-dial = 425
-busy = 425/350,0/350
-ring = 425/1000,0/4000
-congestion = 425/175,0/175
-callwaiting = 425/200,0/5000
-record = 1400/400,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-dialrecall = 425/400,0/40
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[se]
-description = Sweden
-ringcadence = 1000,5000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/5000
-congestion = 425/250,0/750
-callwaiting = 425/200,0/500,425/200,0/9100
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-record = 1400/500,0/15000
-info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; stutter = 425/320,0/20        ; Real swedish standard, not used for now
-
-[sg]
-description = Singapore
-; Singapore
-; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
-; Frequency specs are:   425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
-ringcadence = 400,200,400,2000
-dial        = 425
-ring        = 425*24/400,0/200,425*24/400,0/2000     ; modulation should be 100%, not 90%
-busy        = 425/750,0/750
-congestion  = 425/250,0/250
-callwaiting = 425*24/300,0/200,425*24/300,0/3200
-stutter     = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
-info        = 950/330,1400/330,1800/330,0/1000       ; not currently in use acc. to reference
-dialrecall  = 425*24/500,0/500,425/500,0/2500        ; unspecified in IDA reference, use repeating Holding Tone A,B
-record      = 1400/500,0/15000                       ; unspecified in IDA reference, use 0.5s tone every 15s
-; additionally defined in reference
-nutone      = 425/2500,0/500
-intrusion   = 425/250,0/2000
-warning     = 425/624,0/4376                         ; end of period tone, warning
-acceptance  = 425/125,0/125
-holdinga    = !425*24/500,!0/500                     ; followed by holdingb
-holdingb    = !425/500,!0/2500
-
-[th]
-description = Thailand
-ringcadence = 1000,4000
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-dial = 400*50
-busy = 400/500,0/500
-ring = 420/1000,0/5000
-congestion = 400/300,0/300
-callwaiting = 1000/400,10000/400,1000/400
-; DIALRECALL - not specified - use special dial tone instead.
-dialrecall = 400*50/400,0/100,400*50/400,0/100
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO - specified as an announcement - use special information tones instead
-info = 950/330,1400/330,1800/330
-; STUTTER - not specified
-stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400
-
-[uk]
-description = United Kingdom
-ringcadence = 400,200,400,2000
-; These are the official tones taken from BT SIN350. The actual tones
-; used by BT include some volume differences so sound slightly different
-; from Asterisk-generated ones.
-dial = 350+440
-; Special dial is the intermittent dial tone heard when, for example,
-; you have a divert active on the line
-specialdial = 350+440/750,440/750
-; Busy is also called "Engaged"
-busy = 400/375,0/375
-; "Congestion" is the Beep-bip engaged tone
-congestion = 400/400,0/350,400/225,0/525
-; "Special Congestion" is not used by BT very often if at all
-specialcongestion = 400/200,1004/300
-unobtainable = 400
-ring = 400+450/400,0/200,400+450/400,0/2000
-callwaiting = 400/100,0/4000
-; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
-specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
-; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
-; is and I've not used a payphone for years)
-creditexpired = 400/125,0/125
-; These two are used to confirm/reject service requests on exchanges that
-; don't do voice announcements.
-confirm = 1400
-switching = 400/200,0/400,400/2000,0/400
-; This is the three rising tones Doo-dah-dee "Special Information Tone",
-; usually followed by the BT woman saying an appropriate message.
-info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
-; Not listed in SIN350
-record = 1400/500,0/60000
-stutter = 350+440/750,440/750
-
-[us]
-description = United States / North America
-ringcadence = 2000,4000
-dial = 350+440
-busy = 480+620/500,0/500
-ring = 440+480/2000,0/4000
-congestion = 480+620/250,0/250
-callwaiting = 440/300,0/10000
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[us-old]
-description = United States Circa 1950/ North America
-ringcadence = 2000,4000
-dial = 600*120
-busy = 500*100/500,0/500
-ring = 420*40/2000,0/4000
-congestion = 500*100/250,0/250
-callwaiting = 440/300,0/10000
-dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
-
-[tw]
-description = Taiwan
-; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
-; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
-; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
-; http://www.pbx-manufacturer.com/ky120dx.htm
-; http://www.nettwerked.net/tones.txt
-; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
-;
-; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
-; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
-; ringing tone 440+480Hz 1 sec. on ,2 sec. off
-;
-ringcadence = 1000,4000
-dial = 350+440
-busy = 480+620/500,0/500
-ring = 440+480/1000,0/2000
-congestion = 480+620/250,0/250
-callwaiting = 350+440/250,0/250,350+440/250,0/3250
-dialrecall = 300/1500,0/500
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[ve]
-; Tone definition source for ve found on
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-description = Venezuela / South America
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 400+450/300,0/6000
-dialrecall = 425
-record = 1400/500,0/15000
-info = !950/330,!1440/330,!1800/330,0/1000
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-
-[za]
-description = South Africa
-; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
-; (definitions for other countries can also be found there)
-; Note, though, that South Africa uses two switch types in their network --
-; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
-; The former use 383+417 in dial, ringback etc.  The latter use 400*33
-; I've provided both, uncomment the ones you prefer
-ringcadence = 400,200,400,2000
-; dial/ring/callwaiting for the Siemens switches:
-dial = 400*33
-ring = 400*33/400,0/200,400*33/400,0/2000
-callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
-; dial/ring/callwaiting for the Alcatel switches:
-; dial = 383+417
-; ring = 383+417/400,0/200,383+417/400,0/2000
-; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
-congestion = 400/250,0/250
-busy = 400/500,0/500
-dialrecall = 350+440
-; XXX Not sure about the RECORDTONE
-record = 1400/500,0/10000
-info = 950/330,1400/330,1800/330,0/330
-stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33

version_1.8-reactNative/pabx/etc/asterisk/keys-old/asterisk.crt

@@ -1,23 +0,0 @@
------BEGIN CERTIFICATE-----
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-xbaXOx0EYL/Qzy54aVFrWNEjMPvbRHsnnQ==
------END CERTIFICATE-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/asterisk.csr

@@ -1,16 +0,0 @@
------BEGIN CERTIFICATE REQUEST-----
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-eDQmePUs5Q==
------END CERTIFICATE REQUEST-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/asterisk.key

@@ -1,27 +0,0 @@
------BEGIN RSA PRIVATE KEY-----
-MIIEpQIBAAKCAQEAukC1gWeagfVGJIhrmYl9KsJ2F9TuAzuDzbPTSAH6mk0BVCUa
-hUeUs4bQdsLW8QAV7RC0j4jT/pOKr4xR/OHNsYTfa97XqZXrxI5Wn3Zw6JR2fKek
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-1EdhuPGCHEcQeb2oEhDHKkvgTgHSjWMfL/UtHE2CblofGOYqRqQ29S8=
------END RSA PRIVATE KEY-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/asterisk.pem

@@ -1,50 +0,0 @@
------BEGIN RSA PRIVATE KEY-----
-MIIEpQIBAAKCAQEAukC1gWeagfVGJIhrmYl9KsJ2F9TuAzuDzbPTSAH6mk0BVCUa
-hUeUs4bQdsLW8QAV7RC0j4jT/pOKr4xR/OHNsYTfa97XqZXrxI5Wn3Zw6JR2fKek
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------END RSA PRIVATE KEY-----
------BEGIN CERTIFICATE-----
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------END CERTIFICATE-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/ca.cfg

@@ -1,10 +0,0 @@
-[req]
-distinguished_name = req_distinguished_name
-prompt = no
-
-[req_distinguished_name]
-CN=Asterisk Private CA
-O=Simples IP
-
-[ext]
-basicConstraints=CA:TRUE

version_1.8-reactNative/pabx/etc/asterisk/keys-old/ca.crt

@@ -1,29 +0,0 @@
------BEGIN CERTIFICATE-----
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------END CERTIFICATE-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/ca.key

@@ -1,54 +0,0 @@
------BEGIN RSA PRIVATE KEY-----
-Proc-Type: 4,ENCRYPTED
-DEK-Info: DES-EDE3-CBC,C917EE2EC9629851
-
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------END RSA PRIVATE KEY-----

version_1.8-reactNative/pabx/etc/asterisk/keys-old/tmp.cfg

@@ -1,8 +0,0 @@
-[req]
-distinguished_name = req_distinguished_name
-prompt = no
-
-[req_distinguished_name]
-CN=192.168.88.105
-O=Simples IP
-

version_1.8-reactNative/pabx/etc/asterisk/keys/asterisk.crt

@@ -1,23 +0,0 @@
------BEGIN CERTIFICATE-----
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------END CERTIFICATE-----

version_1.8-reactNative/pabx/etc/asterisk/keys/asterisk.csr

@@ -1,16 +0,0 @@
------BEGIN CERTIFICATE REQUEST-----
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------END CERTIFICATE REQUEST-----