diff --git a/version_1.9.1/README.md b/version_1.9.1/README.md index 678617a..0feff87 100644 --- a/version_1.9.1/README.md +++ b/version_1.9.1/README.md @@ -1,3 +1,7 @@ +# Configuração do PABX versão 1.9.1 no docker + +Inicie a aplicação PABX de maneira fácil e sincronizada. + # Usando PABX no Docker Inicie a aplicação PABX de maneira fácil e sincronizada. @@ -45,8 +49,8 @@ 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. +- pasta_do_postgresql - onde deverá sincronizar o banco de dados. +- id_user - id do usuário sem a necessidade de root para acessar o sistema ### Execute o container com o docker-compose: ```bash @@ -63,23 +67,29 @@ Esse comando tentará compilar e levantar container. Serve para definir variáveis a serem usadas no compose. Leia o arquivo '.env-example' para mais informações. +* ### Banco de dados do PABX 1.9.1 ? +Depois do container inicializar e conseguir acessar o página do PABX execute a página :8080/config-bd.php. Esse script irá modificar o banco para a versão pabx 1.9. O banco de dados é inicializado com a versão 1.8. O processo poderá demorar um pouco. + +* ### Recriar o banco de dados 1.9.1 +Para recriar o banco de dados 1.9. Exclua os conteúdos dos arquivos que aponta a variável `pasta_do_postgresql`. A reinicialização do docker compose irá recriar o banco de dados. Depois que a página de login foi visualizada execute `:8080/config-bd.php` para o banco for definido para 1.9 + * ### 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/supervisor/supervisord.conf` - sem sincronização no arquivo Sincronizados: + - `etc/asterisk` - sincronizado - `/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. @@ -97,4 +107,5 @@ Serve para definir variáveis a serem usadas no compose. Leia o arquivo '.env-ex - `10000-10030` - asterisk (RTP) * ### Se o pabx criar arquivos que não estão no projeto gitea? - Coloque em `.gitignore` e mande a correção. \ No newline at end of file + Coloque em `.gitignore` e mande o PR. + diff --git a/version_1.9.1/docker-compose.yml b/version_1.9.1/docker-compose.yml index 06c4273..5bc100c 100644 --- a/version_1.9.1/docker-compose.yml +++ b/version_1.9.1/docker-compose.yml @@ -55,6 +55,7 @@ services: links: - postgres volumes: + - ./pabx/etc/asterisk/:/etc/asterisk - ${pasta_do_projeto}:/var/www/html/aplicativo - ${pasta_do_projeto}/include:/var/www/html/include - ${pasta_do_projeto}/asterisk/var_lib_asterisk/:/var/lib/asterisk/ diff --git a/version_1.9.1/env_example b/version_1.9.1/env_example index 1e0f3de..89faa8d 100644 --- a/version_1.9.1/env_example +++ b/version_1.9.1/env_example @@ -5,7 +5,7 @@ # Exemplo: #pasta_do_projeto=/home/user/source_pabx #pasta_do_projeto=c://Usuario/User/source_pabx -pasta_do_projeto="/home/rodgger/projetos_simplesip/pabx-app/" +pasta_do_projeto="" # Pasta do host para sincronizar a pasta /var/lib/pgsql/data (versão 13) @@ -14,7 +14,7 @@ pasta_do_projeto="/home/rodgger/projetos_simplesip/pabx-app/" # # Essa pasta pode ser enviado ao repositório caso queira compartilhar # o banco -pasta_do_postgresql="/home/rodgger/projetos_simplesip/bd_1.8" +pasta_do_postgresql="" # ID do usuário para ter acesso ao volume diff --git a/version_1.9.1/pabx/Dockerfile b/version_1.9.1/pabx/Dockerfile index e09b57b..27f865b 100755 --- a/version_1.9.1/pabx/Dockerfile +++ b/version_1.9.1/pabx/Dockerfile @@ -27,7 +27,6 @@ COPY build-asterisk_php.sh / RUN chmod 755 /build-asterisk_php.sh RUN /build-asterisk_php.sh -COPY etc_asterisk/* /etc/asterisk/ # definir as variáveis de ambiente para o cron RUN printenv | sed 's/^\(.*\)$/\1/g' > /etc/environment diff --git a/version_1.9.1/pabx/etc_asterisk/acl.conf b/version_1.9.1/pabx/etc_asterisk/acl.conf deleted file mode 100644 index b052606..0000000 --- a/version_1.9.1/pabx/etc_asterisk/acl.conf +++ /dev/null @@ -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 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: -; -; [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: -; -; [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 diff --git a/version_1.9.1/pabx/etc_asterisk/adsi.conf b/version_1.9.1/pabx/etc_asterisk/adsi.conf deleted file mode 100644 index 0f36f80..0000000 --- a/version_1.9.1/pabx/etc_asterisk/adsi.conf +++ /dev/null @@ -1,8 +0,0 @@ -; -; Sample ADSI Configuration file -; -[intro] -alignment = center -greeting => Welcome to the -greeting => Asterisk -greeting => Open Source PBX diff --git a/version_1.9.1/pabx/etc_asterisk/adtranvofr.conf b/version_1.9.1/pabx/etc_asterisk/adtranvofr.conf deleted file mode 100644 index dc7bcfc..0000000 --- a/version_1.9.1/pabx/etc_asterisk/adtranvofr.conf +++ /dev/null @@ -1,39 +0,0 @@ -; -; Voice over Frame Relay (Adtran style) -; -; Configuration file - -[interfaces] -; -; Default language -; -;language=en -; -; Lines for which we are the user termination. They accept incoming -; and outgoing calls. We use the default context on the first 8 lines -; used by internal phones. -; -context=default -;user => voice00 -;user => voice01 -;user => voice02 -;user => voice03 -;user => voice04 -;user => voice05 -;user => voice06 -;user => voice07 -; Calls on 16 and 17 come from the outside world, so they get -; a little bit special treatment -context=remote -;user => voice16 -;user => voice17 -; -; Next we have lines which we only accept calls on, and typically -; do not send outgoing calls on (i.e. these are where we are the -; network termination) -; -;network => voice08 -;network => voice09 -;network => voice10 -;network => voice11 -;network => voice12 diff --git a/version_1.9.1/pabx/etc_asterisk/agents.conf b/version_1.9.1/pabx/etc_asterisk/agents.conf deleted file mode 100644 index 7a045d3..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents.conf +++ /dev/null @@ -1,6 +0,0 @@ -[general] -#include agents_general.conf -#include agents_general_customizado.conf -#include agents_adicional.conf -#include agents_usuarios.conf -#include agents_usuarios_customizado.conf diff --git a/version_1.9.1/pabx/etc_asterisk/agents_adicional.conf b/version_1.9.1/pabx/etc_asterisk/agents_adicional.conf deleted file mode 100644 index 5567d84..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents_adicional.conf +++ /dev/null @@ -1,16 +0,0 @@ -[agents](!) -;maxlogintries=5 -autologoff=0 -;autologoffunavail=no -ackcall=no -;endcall=no -wrapuptime=0 -musiconhold=agentes -;goodbye= -;updatecdr=no -;group=1 -recordagentcalls=no -;recordformat= -;urlprefix= -;savecallsin= -custom_beep=beep diff --git a/version_1.9.1/pabx/etc_asterisk/agents_general.conf b/version_1.9.1/pabx/etc_asterisk/agents_general.conf deleted file mode 100644 index 9f668a3..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents_general.conf +++ /dev/null @@ -1,2 +0,0 @@ -;persistentagents=no -;multiplelogin=no diff --git a/version_1.9.1/pabx/etc_asterisk/agents_general_customizado.conf b/version_1.9.1/pabx/etc_asterisk/agents_general_customizado.conf deleted file mode 100644 index 8b13789..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents_general_customizado.conf +++ /dev/null @@ -1 +0,0 @@ - diff --git a/version_1.9.1/pabx/etc_asterisk/agents_usuarios.conf b/version_1.9.1/pabx/etc_asterisk/agents_usuarios.conf deleted file mode 100644 index 61b3a56..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents_usuarios.conf +++ /dev/null @@ -1,4 +0,0 @@ -[1000](agents) -fullname=admin -[1001](agents) -fullname=administrador diff --git a/version_1.9.1/pabx/etc_asterisk/agents_usuarios_customizado.conf b/version_1.9.1/pabx/etc_asterisk/agents_usuarios_customizado.conf deleted file mode 100644 index 8b13789..0000000 --- a/version_1.9.1/pabx/etc_asterisk/agents_usuarios_customizado.conf +++ /dev/null @@ -1 +0,0 @@ - diff --git a/version_1.9.1/pabx/etc_asterisk/alarmreceiver.conf b/version_1.9.1/pabx/etc_asterisk/alarmreceiver.conf deleted file mode 100644 index e4815a9..0000000 --- a/version_1.9.1/pabx/etc_asterisk/alarmreceiver.conf +++ /dev/null @@ -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 -; diff --git a/version_1.9.1/pabx/etc_asterisk/alsa.conf b/version_1.9.1/pabx/etc_asterisk/alsa.conf deleted file mode 100644 index 3e61710..0000000 --- a/version_1.9.1/pabx/etc_asterisk/alsa.conf +++ /dev/null @@ -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". -; ---------------------------------------------------------------------------------- diff --git a/version_1.9.1/pabx/etc_asterisk/amd.conf b/version_1.9.1/pabx/etc_asterisk/amd.conf deleted file mode 100644 index 84b391c..0000000 --- a/version_1.9.1/pabx/etc_asterisk/amd.conf +++ /dev/null @@ -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. diff --git a/version_1.9.1/pabx/etc_asterisk/app_mysql.conf b/version_1.9.1/pabx/etc_asterisk/app_mysql.conf deleted file mode 100644 index fafd4f7..0000000 --- a/version_1.9.1/pabx/etc_asterisk/app_mysql.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/app_skel.conf b/version_1.9.1/pabx/etc_asterisk/app_skel.conf deleted file mode 100644 index ada8461..0000000 --- a/version_1.9.1/pabx/etc_asterisk/app_skel.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/applyzap.conf b/version_1.9.1/pabx/etc_asterisk/applyzap.conf deleted file mode 100644 index e69de29..0000000 diff --git a/version_1.9.1/pabx/etc_asterisk/ari.conf b/version_1.9.1/pabx/etc_asterisk/ari.conf deleted file mode 100644 index 3d0ad55..0000000 --- a/version_1.9.1/pabx/etc_asterisk/ari.conf +++ /dev/null @@ -1,30 +0,0 @@ -[general] -enabled = no ; 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 diff --git a/version_1.9.1/pabx/etc_asterisk/ast_debug_tools.conf b/version_1.9.1/pabx/etc_asterisk/ast_debug_tools.conf deleted file mode 100644 index f26626b..0000000 --- a/version_1.9.1/pabx/etc_asterisk/ast_debug_tools.conf +++ /dev/null @@ -1,57 +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)) - -# 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 "/" -# format or in abbreviation format such as "CST6CDT". If not -# specified, the "local" timezone is used. -# LOG_TIMEZONE= diff --git a/version_1.9.1/pabx/etc_asterisk/asterisk.adsi b/version_1.9.1/pabx/etc_asterisk/asterisk.adsi deleted file mode 100644 index 904b33a..0000000 --- a/version_1.9.1/pabx/etc_asterisk/asterisk.adsi +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/asterisk.conf b/version_1.9.1/pabx/etc_asterisk/asterisk.conf deleted file mode 100644 index fa9cfab..0000000 --- a/version_1.9.1/pabx/etc_asterisk/asterisk.conf +++ /dev/null @@ -1,114 +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 = 45 -;debug = 3 -;alwaysfork = yes ; Same as -F at startup. -;nofork = yes ; Same as -f at startup. -;quiet = yes ; Same as -q at startup. -;timestamp = yes ; Same as -T at startup. -;execincludes = yes ; Support #exec in config files. -;console = yes ; Run as console (same as -c at startup). -;highpriority = yes ; Run realtime priority (same as -p at - ; startup). -;initcrypto = yes ; Initialize crypto keys (same as -i at - ; startup). -;nocolor = yes ; Disable console colors. -;dontwarn = yes ; Disable some warnings. -;dumpcore = yes ; Dump core on crash (same as -g at startup). -;languageprefix = yes ; Use the new sound prefix path syntax. -;systemname = my_system_name ; Prefix uniqueid with a system name for - ; Global uniqueness issues. -;autosystemname = yes ; Automatically set systemname to hostname, - ; uses 'localhost' on failure, or systemname if - ; set. -;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms) - ; If we get shorter DTMF messages, these will be - ; changed to the minimum duration -;maxcalls = 10 ; Maximum amount of calls allowed. -;maxload = 0.9 ; Asterisk stops accepting new calls if the - ; load average exceed this limit. -;maxfiles = 1000 ; Maximum amount of openfiles. -;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if - ; the amount of free memory falls below this - ; watermark. -;cache_record_files = yes ; Cache recorded sound files to another - ; directory during recording. -;record_cache_dir = /tmp ; Specify cache directory (used in conjunction - ; with cache_record_files). -;transmit_silence = yes ; Transmit silence while a channel is in a - ; waiting state, a recording only state, or - ; when DTMF is being generated. Note that the - ; silence internally is generated in raw signed - ; linear format. This means that it must be - ; transcoded into the native format of the - ; channel before it can be sent to the device. - ; It is for this reason that this is optional, - ; as it may result in requiring a temporary - ; codec translation path for a channel that may - ; not otherwise require one. -;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of - ; directly. -runuser = pbx ; The user to run as. -rungroup = pbx ; The group to run as. -;lightbackground = yes ; If your terminal is set for a light-colored - ; background. -;forceblackbackground = yes ; Force the background of the terminal to be - ; black, in order for terminal colors to show - ; up properly. -;defaultlanguage = en ; Default language -documentation_language = en_US ; Set the language you want documentation - ; displayed in. Value is in the same format as - ; locale names. -;hideconnect = yes ; Hide messages displayed when a remote console - ; connects and disconnects. -;lockconfdir = no ; Protect the directory containing the - ; configuration files (/etc/asterisk) with a - ; lock. -;stdexten = gosub ; How to invoke the extensions.conf stdexten. - ; macro - Invoke the stdexten using a macro as - ; done by legacy Asterisk versions. - ; gosub - Invoke the stdexten using a gosub as - ; documented in extensions.conf.sample. - ; Default gosub. -;live_dangerously = no ; Enable the execution of 'dangerous' dialplan - ; functions from external sources (AMI, - ; etc.) These functions (such as SHELL) are - ; considered dangerous because they can allow - ; privilege escalation. - ; Default no -;entityid=00:11:22:33:44:55 ; Entity ID. - ; This is in the form of a MAC address. - ; It should be universally unique. - ; It must be unique between servers communicating - ; with a protocol that uses this value. - ; This is currently is used by DUNDi and - ; Exchanging Device and Mailbox State - ; using protocols: XMPP, Corosync and PJSIP. -;rtp_pt_dynamic = 96 ; Normally the Dynamic RTP Payload Type numbers - ; are 96-127, which allow 32 formats. When you - ; use more and receive the message "No Dynamic - ; RTP mapping available", extend the dynamic - ; range by going for 35 (or 0) instead of 96. - ; This allows 29 (or 64) more formats. 96 is the - ; default because any number below might be - ; rejected by a remote implementation; although - ; no such broken implementation is known, yet. - -; Changing the following lines may compromise your security. -;[files] -;astctlpermissions = 0660 -;astctlowner = root -;astctlgroup = apache -;astctl = asterisk.ctl diff --git a/version_1.9.1/pabx/etc_asterisk/calendar.conf b/version_1.9.1/pabx/etc_asterisk/calendar.conf deleted file mode 100644 index 8c73361..0000000 --- a/version_1.9.1/pabx/etc_asterisk/calendar.conf +++ /dev/null @@ -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 ${CALENDAR_EVENT()} dialplan function: -; -; summary : The VEVENT Summary property or Exchange subject -; description : The text description of the vent -; organizer : The organizer of the event -; location : The location field of the event -; calendar : The name of the calendar tied to the event -; uid : The unique ID for this event -; start : Start time of the event -; end : The end time of the event -; busystate : 0=FREE, 1=TENTATIVE, 2=BUSY -; -;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins) -; -;channel = SIP/60001 ; Channel to dial -;context = default ; Context to connect to on answer -;extension = 123 ; Extension to connect to on answer -; -; or -; -;app = Playback ; Application to execute on answer (instead of context/extension) -;appdata = tt-weasels ; Data part of application to execute on answer -; -;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds -; -; Channel variables can be set on the notification channel. The format is -; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan -; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value -; of earlier variables in the definition of later ones. -; -;setvar = CALLERID(name)=${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 diff --git a/version_1.9.1/pabx/etc_asterisk/ccss.conf b/version_1.9.1/pabx/etc_asterisk/ccss.conf deleted file mode 100644 index 7b3fe7d..0000000 --- a/version_1.9.1/pabx/etc_asterisk/ccss.conf +++ /dev/null @@ -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= diff --git a/version_1.9.1/pabx/etc_asterisk/cdr.conf b/version_1.9.1/pabx/etc_asterisk/cdr.conf deleted file mode 100644 index cb3f24c..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr.conf +++ /dev/null @@ -1,167 +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 -unanswered = yes - -; Define whether or not to log congested calls. Setting this to "yes" will -; report each call that fails to complete due to congestion conditions. Default -; is "no". -;congestion = no - -; Normally, CDR's are not closed out until after all extensions are finished -; executing. By enabling this option, the CDR will be ended before executing -; the "h" extension and hangup handlers so that CDR values such as "end" and -; "billsec" may be retrieved inside of of this extension. -; The default value is "no". -;endbeforehexten=no - -; Normally, the 'billsec' field logged to the backends (text files or databases) -; is simply the end time (hangup time) minus the answer time in seconds. Internally, -; asterisk stores the time in terms of microseconds and seconds. By setting -; initiatedseconds to 'yes', you can force asterisk to report any seconds -; that were initiated (a sort of round up method). Technically, this is -; when the microsecond part of the end time is greater than the microsecond -; part of the answer time, then the billsec time is incremented one second. -; The default value is "no". -;initiatedseconds=no - -; Define the CDR batch mode, where instead of posting the CDR at the end of -; every call, the data will be stored in a buffer to help alleviate load on the -; asterisk server. Default is "no". -; -; WARNING WARNING WARNING -; Use of batch mode may result in data loss after unsafe asterisk termination -; ie. software crash, power failure, kill -9, etc. -; WARNING WARNING WARNING -; -;batch=no - -; Define the maximum number of CDRs to accumulate in the buffer before posting -; them to the backend engines. 'batch' must be set to 'yes'. Default is 100. -;size=100 - -; Define the maximum time to accumulate CDRs in the buffer before posting them -; to the backend engines. If this time limit is reached, then it will post the -; records, regardless of the value defined for 'size'. 'batch' must be set to -; 'yes'. Note that time is in seconds. Default is 300 (5 minutes). -;time=300 - -; The CDR engine uses the internal asterisk scheduler to determine when to post -; records. Posting can either occur inside the scheduler thread, or a new -; thread can be spawned for the submission of every batch. For small batches, -; it might be acceptable to just use the scheduler thread, so set this to "yes". -; For large batches, say anything over size=10, a new thread is recommended, so -; set this to "no". Default is "no". -;scheduleronly=no - -; When shutting down asterisk, you can block until the CDRs are submitted. If -; you don't, then data will likely be lost. You can always check the size of -; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on -; submission of CDR data during asterisk shutdown, set this to "yes". Default -; is "yes". -;safeshutdown=yes - -; -; -; CHOOSING A CDR "BACKEND" (what kind of output to generate) -; -; To choose a backend, you have to make sure either the right category is -; defined in this file, or that the appropriate config file exists, and has the -; proper definitions in it. If there are any problems, usually, the entry will -; silently ignored, and you get no output. -; -; Also, please note that you can generate CDR records in as many formats as you -; wish. If you configure 5 different CDR formats, then each event will be logged -; in 5 different places! In the example config files, all formats are commented -; out except for the cdr-csv format. -; -; Here are all the possible back ends: -; -; csv, custom, manager, odbc, pgsql, radius, sqlite, tds -; (also, mysql is available via the asterisk-addons, due to licensing -; requirements) -; (please note, also, that other backends can be created, by creating -; a new backend module in the source cdr/ directory!) -; -; Some of the modules required to provide these backends will not build or install -; unless some dependency requirements are met. Examples of this are pgsql, odbc, -; etc. If you are not getting output as you would expect, the first thing to do -; is to run the command "make menuselect", and check what modules are available, -; by looking in the "2. Call Detail Recording" option in the main menu. If your -; backend is marked with XXX, you know that the "configure" command could not find -; the required libraries for that option. -; -; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv -; file, define the [csv] category in this file. No database necessary. The example -; config files are set up to provide this kind of output by default. -; -; To get custom csv CDR records, make sure the cdr_custom.conf file -; is present, and contains the proper [mappings] section. The advantage to -; using this backend, is that you can define which fields to output, and in -; what order. By default, the example configs are set up to mimic the cdr-csv -; output. If you don't make any changes to the mappings, you are basically generating -; the same thing as cdr-csv, but expending more CPU cycles to do so! -; -; To get manager events generated, make sure the cdr_manager.conf file exists, -; and the [general] section is defined, with the single variable 'enabled = yes'. -; -; For odbc, make sure all the proper libs are installed, that "make menuselect" -; shows that the modules are available, and the cdr_odbc.conf file exists, and -; has a [global] section with the proper variables defined. -; -; For pgsql, make sure all the proper libs are installed, that "make menuselect" -; shows that the modules are available, and the cdr_pgsql.conf file exists, and -; has a [global] section with the proper variables defined. -; -; For logging to radius databases, make sure all the proper libs are installed, that -; "make menuselect" shows that the modules are available, and the [radius] -; category is defined in this file, and in that section, make sure the 'radiuscfg' -; variable is properly pointing to an existing radiusclient.conf file. -; -; For logging to sqlite databases, make sure the 'cdr.db' file exists in the log directory, -; which is usually /var/log/asterisk. Of course, the proper libraries should be available -; during the 'configure' operation. -; -; For tds logging, make sure the proper libraries are available during the 'configure' -; phase, and that cdr_tds.conf exists and is properly set up with a [global] category. -; -; Also, remember, that if you wish to log CDR info to a database, you will have to define -; a specific table in that databse to make things work! See the doc directory for more details -; on how to create this table in each database. -; - -[csv] -usegmtime=yes ; log date/time in GMT. Default is "no" -loguniqueid=yes ; log uniqueid. Default is "no" -loguserfield=yes ; log user field. Default is "no" -accountlogs=yes ; create separate log file for each account code. Default is "yes" -newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns (peeraccount, linkedid, sequence). - ; Default is "no". - -;[radius] -;usegmtime=yes ; log date/time in GMT -;loguniqueid=yes ; log uniqueid -;loguserfield=yes ; log user field -; Set this to the location of the radiusclient-ng configuration file -; The default is /etc/radiusclient-ng/radiusclient.conf -;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_adaptive_odbc.conf b/version_1.9.1/pabx/etc_asterisk/cdr_adaptive_odbc.conf deleted file mode 100644 index 2daa983..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_adaptive_odbc.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_custom.conf b/version_1.9.1/pabx/etc_asterisk/cdr_custom.conf deleted file mode 100644 index a7bf0f6..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_custom.conf +++ /dev/null @@ -1,16 +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)})},${CDR(sequence)},${CSV_QUOTE(${CDR(direcao)})},${CSV_QUOTE(${CDR(ramal_origem)})},${CSV_QUOTE(${CDR(fora_horario)})} - - -; -; High Resolution Time for billsec and duration fields -;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)} -;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})} diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_manager.conf b/version_1.9.1/pabx/etc_asterisk/cdr_manager.conf deleted file mode 100644 index b95038c..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_manager.conf +++ /dev/null @@ -1,6 +0,0 @@ -; -; Asterisk Call Management CDR -; -[general] -enabled = yes - diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_mysql.conf b/version_1.9.1/pabx/etc_asterisk/cdr_mysql.conf deleted file mode 100644 index a1f7d38..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_mysql.conf +++ /dev/null @@ -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= -;ssl_cert= -;ssl_key= -; -; You may also configure the field names used in the CDR table. -; -[columns] -;static "" => -;alias => -alias start => calldate -;alias clid => -;alias src => -;alias dst => -;alias dcontext => -;alias channel => -;alias dstchannel => -;alias lastapp => -;alias lastdata => -;alias duration => -;alias billsec => -;alias disposition => -;alias amaflags => -;alias accountcode => -;alias userfield => -;alias uniqueid => diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_odbc.conf b/version_1.9.1/pabx/etc_asterisk/cdr_odbc.conf deleted file mode 100644 index 663ce09..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_odbc.conf +++ /dev/null @@ -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) diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_pgsql.conf b/version_1.9.1/pabx/etc_asterisk/cdr_pgsql.conf deleted file mode 100644 index 830ace7..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_pgsql.conf +++ /dev/null @@ -1,8 +0,0 @@ -[global] -hostname=127.0.0.1 -port=5432 -dbname=pbx -password=ctepgSQL -user=contacte -table=ast_bilhetes -encoding=LATIN1 diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_sqlite3_custom.conf b/version_1.9.1/pabx/etc_asterisk/cdr_sqlite3_custom.conf deleted file mode 100644 index 0d5dc09..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_sqlite3_custom.conf +++ /dev/null @@ -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)}' diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_syslog.conf b/version_1.9.1/pabx/etc_asterisk/cdr_syslog.conf deleted file mode 100644 index 3a619be..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_syslog.conf +++ /dev/null @@ -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)}" diff --git a/version_1.9.1/pabx/etc_asterisk/cdr_tds.conf b/version_1.9.1/pabx/etc_asterisk/cdr_tds.conf deleted file mode 100644 index f3a9d7c..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cdr_tds.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/cel.conf b/version_1.9.1/pabx/etc_asterisk/cel.conf deleted file mode 100644 index 344a8d7..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel.conf +++ /dev/null @@ -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=no - -; Application Tracking -; -; Use the 'apps' keyword to specify the list of applications for which you want -; to receive CEL events. This is a comma separated list of Asterisk dialplan -; applications, such as Dial, Queue, and Park. -; -; Accepted values: A comma separated list of Asterisk dialplan applications -; Default value: none -; -; Note: You may also use 'all' which will result in CEL events being reported -; for all Asterisk applications. This may affect Asterisk's performance -; significantly. - -apps=dial,park - -; Event Tracking -; -; Use the 'events' keyword to specify the list of events which you want to be -; raised when they occur. This is a comma separated list of the values in the -; table below. -; -; Accepted values: A comma separated list of one or more of the following: -; ALL -- Generate entries on all events -; CHAN_START -- The time a channel was created -; CHAN_END -- The time a channel was terminated -; ANSWER -- The time a channel was answered (ie, phone taken off-hook) -; HANGUP -- The time at which a hangup occurred -; BRIDGE_ENTER -- The time a channel was connected into a conference room -; BRIDGE_EXIT -- The time a channel was removed from a conference room -; APP_START -- The time a tracked application was started -; APP_END -- the time a tracked application ended -; PARK_START -- The time a call was parked -; PARK_END -- Unpark event -; BLINDTRANSFER -- When a blind transfer is initiated -; ATTENDEDTRANSFER -- When an attended transfer is initiated -; PICKUP -- This channel picked up the specified channel -; FORWARD -- This channel is being forwarded somewhere else -; LINKEDID_END -- The last channel with the given linkedid is retired -; USER_DEFINED -- Triggered from the dialplan, and has a name given by the -; user -; LOCAL_OPTIMIZE -- A local channel pair is optimizing away. -; -; Default value: none -; (Track no events) - -events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT - -; Date Format -; -; Use the 'dateformat' keyword to specify the date format used when CEL events -; are raised. -; -; Accepted values: A strftime format string (see man strftime) -; -; Example: "%F %T" -; -> This gives the date and time in the format "2009-06-23 17:02:35" -; -; If this option is not specified, the default format is "." -; 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 -; diff --git a/version_1.9.1/pabx/etc_asterisk/cel_custom.conf b/version_1.9.1/pabx/etc_asterisk/cel_custom.conf deleted file mode 100644 index 8ac9e29..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel_custom.conf +++ /dev/null @@ -1,39 +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})} - - -[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})} diff --git a/version_1.9.1/pabx/etc_asterisk/cel_odbc.conf b/version_1.9.1/pabx/etc_asterisk/cel_odbc.conf deleted file mode 100644 index 0c0b83f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel_odbc.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/cel_pgsql.conf b/version_1.9.1/pabx/etc_asterisk/cel_pgsql.conf deleted file mode 100644 index 42de1a1..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel_pgsql.conf +++ /dev/null @@ -1,93 +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 - -;CREATE TABLE pbx_cel ( -; id bigserial , -; eventtype varchar (32) NOT NULL , -; eventtime timestamp NOT NULL , -; userdeftype varchar(256) NOT NULL , -; cid_name varchar (128) NOT NULL , -; cid_num varchar (128) NOT NULL , -; cid_ani varchar (128) NOT NULL , -; cid_rdnis varchar (128) NOT NULL , -; cid_dnid varchar (128) NOT NULL , -; exten varchar (128) NOT NULL , -; context varchar (128) NOT NULL , -; channame varchar (128) NOT NULL , -; appname varchar (128) NOT NULL , -; appdata varchar (128) NOT NULL , -; amaflags int NOT NULL , -; accountcode varchar (32) NOT NULL , -; peeraccount varchar (32) NOT NULL , -; uniqueid varchar (160) NOT NULL , -; linkedid varchar (160) NOT NULL , -; userfield varchar (256) NOT NULL , -; peer varchar (128) NOT NULL -;); - - -[global] -; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field, -; instead of (by default) just putting the user defined event name there. -; -;show_user_defined=yes - -hostname=127.0.0.1 -port=5432 -dbname=pbx -password=ctepgSQL -user=contacte -table=pbx_cel ;SQL table where CEL's will be inserted -;appname=asterisk ; Postgres application_name support (optional). Whitespace not allowed. \ No newline at end of file diff --git a/version_1.9.1/pabx/etc_asterisk/cel_sqlite3_custom.conf b/version_1.9.1/pabx/etc_asterisk/cel_sqlite3_custom.conf deleted file mode 100644 index 2d9a24f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel_sqlite3_custom.conf +++ /dev/null @@ -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}' diff --git a/version_1.9.1/pabx/etc_asterisk/cel_tds.conf b/version_1.9.1/pabx/etc_asterisk/cel_tds.conf deleted file mode 100644 index 399093b..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cel_tds.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/chan_dahdi.conf b/version_1.9.1/pabx/etc_asterisk/chan_dahdi.conf deleted file mode 100644 index 3362fd6..0000000 --- a/version_1.9.1/pabx/etc_asterisk/chan_dahdi.conf +++ /dev/null @@ -1,7 +0,0 @@ -[trunkgroups] - -[channels] -#include chan_dahdi_troncos.conf -#include chan_dahdi_troncos_general.conf -#include chan_dahdi_ramais_general.conf -#include chan_dahdi_ramais.conf diff --git a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_ramais.conf b/version_1.9.1/pabx/etc_asterisk/chan_dahdi_ramais.conf deleted file mode 100644 index e69de29..0000000 diff --git a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_ramais_general.conf b/version_1.9.1/pabx/etc_asterisk/chan_dahdi_ramais_general.conf deleted file mode 100644 index d89d75f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_ramais_general.conf +++ /dev/null @@ -1,21 +0,0 @@ -;Config. Portas FXS -language=pt_br -usecallerid=yes -callwaiting=no -usecallingpres=yes -callwaitingcallerid=yes -threewaycalling=yes -transfer=yes -canpark=yes -cancallforward=yes -callreturn=yes -;rxgain=3 -;txgain=-7 -rxgain=0 -txgain=-6 -echocancel=128 -echocancelwhenbridged=yes -echotraining=yes -relaxdtmf=yes -rxflash=600 -group=10 diff --git a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_troncos.conf b/version_1.9.1/pabx/etc_asterisk/chan_dahdi_troncos.conf deleted file mode 100644 index e69de29..0000000 diff --git a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_troncos_general.conf b/version_1.9.1/pabx/etc_asterisk/chan_dahdi_troncos_general.conf deleted file mode 100644 index 7fb2dda..0000000 --- a/version_1.9.1/pabx/etc_asterisk/chan_dahdi_troncos_general.conf +++ /dev/null @@ -1,15 +0,0 @@ -language=pt_br -usecallerid=yes -callwaiting=no -usecallingpres=yes -callwaitingcallerid=yes -;rxgain=1 -;txgain=-7 -;rxgain=4 -;txgain=4 -;echocancel=128 -;echocancelwhenbridged=yes -;echotraining=yes -;busydetect=yes -;busycount=4 -relaxdtmf=yes \ No newline at end of file diff --git a/version_1.9.1/pabx/etc_asterisk/chan_mobile.conf b/version_1.9.1/pabx/etc_asterisk/chan_mobile.conf deleted file mode 100644 index 3814337..0000000 --- a/version_1.9.1/pabx/etc_asterisk/chan_mobile.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/cli.conf b/version_1.9.1/pabx/etc_asterisk/cli.conf deleted file mode 100644 index 0ddd92c..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cli.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/cli_aliases.conf b/version_1.9.1/pabx/etc_asterisk/cli_aliases.conf deleted file mode 100644 index adaed90..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cli_aliases.conf +++ /dev/null @@ -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. diff --git a/version_1.9.1/pabx/etc_asterisk/cli_permissions.conf b/version_1.9.1/pabx/etc_asterisk/cli_permissions.conf deleted file mode 100644 index 4a6973f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/cli_permissions.conf +++ /dev/null @@ -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 = | all ; allow the user to run 'command' | -; ; allow the user to run 'all' the commands -; deny = | 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 -; -; diff --git a/version_1.9.1/pabx/etc_asterisk/codecs.conf b/version_1.9.1/pabx/etc_asterisk/codecs.conf deleted file mode 100644 index e1dbd79..0000000 --- a/version_1.9.1/pabx/etc_asterisk/codecs.conf +++ /dev/null @@ -1,206 +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 -genericplc => true - -; Generate custom formats for formats requiring attributes. -; After defining the custom format, the name used in defining -; the format can be used throughout Asterisk in the format 'allow' -; and 'disallow' options. -; -; Example: silk8 is a predefined custom format in this config file. -; Once this config file is loaded, silk8 can be used anywhere a -; peer's codec capabilities are defined. -; -; In sip.conf 'silk8' can be defined as a capability for a peer. -; [peer1] -; type=peer -; host=dynamic -; disallow=all -; allow=silk8 ;custom codec defined in codecs.conf -; -; LIMITATIONS -; Custom formats can only be defined at startup. Any changes to this -; file made after startup will not take into effect until after Asterisk -; is restarted. -; - -; Default Custom SILK format definitions, only one custom SILK format per -; sample rate is allowed. -[silk8] -type=silk -samprate=8000 -fec=true ; turn on or off encoding with forward error correction. - ; On recommended, off by default. -packetloss_percentage=10 ; Estimated packet loss percentage in uplink direction. This - ; affects how much redundancy is built in when using fec. - ; The higher the percentage, the larger amount of bandwidth is - ; used. Default is 0%, 10% is recommended when fec is in use. - -maxbitrate=10000 ; Use the table below to make sure a useful bitrate is choosen - ; for maxbitrate. If not set or value is not within the bounds - ; of the encoder, a default value is chosen. - ; - ; sample rate | bitrate range - ; 8khz | 5000 - 20000 bps - ; 12khz | 7000 - 25000 bps - ; 16khz | 8000 - 30000 bps - ; 24khz | 20000- 40000 bps - ; -;dtx=true ; Encode using discontinuous transmission mode or not. Turning this - ; on will save bandwidth during periods of silence at the cost of - ; increased computational complexity. Off by default. - -[silk12] -type=silk -samprate=12000 -maxbitrate=12000 -fec=true -packetloss_percentage=10; - -[silk16] -type=silk -samprate=16000 -maxbitrate=20000 -fec=true -packetloss_percentage=10; - -[silk24] -type=silk -samprate=24000 -maxbitrate=30000 -fec=true -packetloss_percentage=10; - - -; Default custom CELT codec definitions. Only one custom CELT definition is allowed -; per a sample rate. -;[celt44] -;type=celt -;samprate=44100 ; The samplerate in hz. This option is required. -;framesize=480 ; The framesize option represents the duration of each frame in samples. - ; This must be a factor of 2. This option is only advertised in an SDP - ; when it is set. Otherwise a default of framesize of 480 is assumed - ; internally - -;[celt48] -;type=celt -;samprate=48000 - -;[celt32] -;type=celt -;samprate=32000 - -;============================ OPUS Section Options ============================ -; -;[opus] -;type= ; Must be of type "opus" (default: "") -;packet_loss= ; Encoder's packet loss percentage. Can be any number between 0 - ; and 100, inclusive. A higher value results in more loss - ; resistance. (default: 0) -;complexity= ; Encoder's computational complexity. Can be any number between 0 - ; and 10, inclusive. Note, 10 equals the highest complexity. - ; (default: 10) -;max_bandwidth= ; Encoder's maximum bandwidth allowed. Sets an upper bandwidth - ; bound on the encoder. Can be any of the following: narrow, - ; medium, wide, super_wide, full. (default: full) -;signal= ; Encoder's signal type. Aids in mode selection on the encoder: Can - ; be any of the following: auto, voice, music. (default: auto) -;application= ; Encoder's application type. Can be any of the following: voip, - ; audio, low_delay. (default: voip) -;max_playback_rate= ; Override the maximum playback rate in the offer's SDP. - ; Any value between 8000 and 48000 (inclusive) is valid, - ; however typically it should match one of the usual opus - ; bandwidths. A value of "sdp" is also allowed. When set - ; to "sdp" then the value from the offer's SDP is used. - ; (default: "sdp") -;bitrate= ; Override the maximum average bitrate in the offer's SDP. Any value - ; between 500 and 512000 is valid. The following values are also - ; allowed: auto, max, sdp. When set to "sdp" then the value from - ; the offer's sdp is used. (default: "sdp") -;cbr= ; Override the constant bit rate parameter in the offer's SDP. A value of - ; 0/false/no represents a variable bit rate whereas 1/true/yes represents - ; a constant bit rate. A value of "sdp" is also allowed. When set to "sdp" - ; then the value from the offer's sdp is used. (default: "sdp") -;fec= ; Override the use inband fec parameter in the offer's SDP. A value of - ; 0/false/no represents disabled whereas 1/true/yes represents enabled. - ; A value of "sdp" is also allowed. When set to "sdp" then the value from - ; the offer's sdp is used. (default: "sdp") -;dtx= ; Override the use dtx parameter in the offer's SDP. A value of 0/false/no - ; represents disabled whereas 1/true/yes represents enabled. A value of - ; "sdp" is also allowed. When set to "sdp" then the value from the offer's - ; sdp is used. (default: "sdp") - -;=============================== OPUS Examples ================================ -; -;[opus] -;type=opus -;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder -;fec=no ; Force no inband fec on the encoder (i.e don't use what's on the SDP) - -;[myopus] -;type=opus -;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz -; ; audio bandwidth at 16Khz sample rate) -;cbr=yes ; Force a constant bit rate (i.e don't use what's on the SDP) diff --git a/version_1.9.1/pabx/etc_asterisk/confbridge.conf b/version_1.9.1/pabx/etc_asterisk/confbridge.conf deleted file mode 100644 index f1fccf6..0000000 --- a/version_1.9.1/pabx/etc_asterisk/confbridge.conf +++ /dev/null @@ -1,12 +0,0 @@ -[general] -[default_user_simplesip] -type=user -dtmf_passthrough=no -dsp_silence_threshold=2500 -dsp_drop_silence=yes - -[simplesip_menu] -type=menu -1=dialplan_exec(adiciona-conferencia,1,1) -2=admin_kick_last -9=admin_toggle_mute_participants \ No newline at end of file diff --git a/version_1.9.1/pabx/etc_asterisk/config_test.conf b/version_1.9.1/pabx/etc_asterisk/config_test.conf deleted file mode 100644 index b7cb212..0000000 --- a/version_1.9.1/pabx/etc_asterisk/config_test.conf +++ /dev/null @@ -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] diff --git a/version_1.9.1/pabx/etc_asterisk/console.conf b/version_1.9.1/pabx/etc_asterisk/console.conf deleted file mode 100644 index aad306e..0000000 --- a/version_1.9.1/pabx/etc_asterisk/console.conf +++ /dev/null @@ -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. diff --git a/version_1.9.1/pabx/etc_asterisk/dahdi_guiread.conf b/version_1.9.1/pabx/etc_asterisk/dahdi_guiread.conf deleted file mode 100644 index 0d3831a..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dahdi_guiread.conf +++ /dev/null @@ -1,5 +0,0 @@ - -[general] - -#include "../zaptel.conf" - diff --git a/version_1.9.1/pabx/etc_asterisk/dahdi_scan.conf b/version_1.9.1/pabx/etc_asterisk/dahdi_scan.conf deleted file mode 100644 index 309d3aa..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dahdi_scan.conf +++ /dev/null @@ -1,11 +0,0 @@ -[1] -active=yes -alarms=UNCONFIGURED -description=ZTDUMMY/1 (source: HRtimer) 1 -name=ZTDUMMY/1 -manufacturer= -devicetype=Zaptel Dummy Timing Driver -location= -basechan=1 -totchans=0 -irq=0 diff --git a/version_1.9.1/pabx/etc_asterisk/dbsep.conf b/version_1.9.1/pabx/etc_asterisk/dbsep.conf deleted file mode 100644 index 7a68850..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dbsep.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/digivoice.conf b/version_1.9.1/pabx/etc_asterisk/digivoice.conf deleted file mode 100644 index 89a29df..0000000 --- a/version_1.9.1/pabx/etc_asterisk/digivoice.conf +++ /dev/null @@ -1,27 +0,0 @@ -[general] -allow_slinear=0 -allow_ulaw=1 -allow_alaw=1 -allow_gsm=0 -consolelanguage=br - -[allportsconfig] -default_callprogress=cp_default.cfg -detectiontype=1 -dialtype=1 -afterdialpause=1000 -ringbacktone1=tone1 -ringbacktone2=tone1 -ringbacktimes=1000,4000,1000,4000 -busytone1=tone1 -busytone2=tone1 -busytimes=250,250,250,250 -subchannelringtone1=tone1 -subchannelringtone2=tone1 -subchannelringtimes=100,100,100,5000 - -[groups] - -[port_config] - -[e1_config] \ No newline at end of file diff --git a/version_1.9.1/pabx/etc_asterisk/dnsmgr.conf b/version_1.9.1/pabx/etc_asterisk/dnsmgr.conf deleted file mode 100644 index f028acc..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dnsmgr.conf +++ /dev/null @@ -1,5 +0,0 @@ -[general] -enable=no ; enable creation of managed DNS lookups - ; default is 'no' -;refreshinterval=1200 ; refresh managed DNS lookups every seconds - ; default is 300 (5 minutes) \ No newline at end of file diff --git a/version_1.9.1/pabx/etc_asterisk/dsp.conf b/version_1.9.1/pabx/etc_asterisk/dsp.conf deleted file mode 100644 index f13ca2f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dsp.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/dundi.conf b/version_1.9.1/pabx/etc_asterisk/dundi.conf deleted file mode 100644 index 70f97d4..0000000 --- a/version_1.9.1/pabx/etc_asterisk/dundi.conf +++ /dev/null @@ -1,268 +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 (Should generally be the MAC address of the -; machine it's running on. Defaults to the first eth address, but you -; can override it here, as long as you set it to the MAC of *something* -; you own!) The EID can be overridden by a setting in asterisk.conf, -; or by setting this option. -; -;entityid=00:07:E9:3B:76:60 -; -; Peers shall cache our query responses for the specified time, -; given in seconds. Default is 3600. -; -;cachetime=3600 -; -; This defines the max depth in which to search the DUNDi system. -; Note that the maximum time that we will wait for a response is -; (2000 + 200 * ttl) ms. -; -ttl=32 -; -; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set -; to yes, then we cancel the whole thing (that's enough time for one -; retransmission only). This is used to keep things from stalling for a long -; time for a host that is not available, but would be ill advised for bad -; connections. In addition to 'yes' or 'no' you can also specify a number -; of milliseconds. See 'qualify' for individual peers to turn on for just -; a specific peer. -; -autokill=yes -; -; pbx_dundi creates a rotating key called "secret", under the family -; 'secretpath'. The default family is dundi (resulting in -; the key being held at dundi/secret). -; -;secretpath=dundi -; -; The 'storehistory' option (also changeable at runtime with -; 'dundi store history' and 'dundi no store history') will -; cause the DUNDi engine to keep track of the last several -; queries and the amount of time each query took to execute -; for the purpose of tracking slow nodes. This option is -; off by default due to performance impacts. -; -;storehistory=yes - -[mappings] -; -; The "mappings" section maps DUNDi contexts -; to contexts on the local asterisk system. Remember -; that numbers that are made available under the e164 -; DUNDi context are regulated by the DUNDi General Peering -; Agreement (GPA) if you are a member of the DUNDi E.164 -; Peering System. -; -; dundi_context => local_context,weight,tech,dest[,options]] -; -; 'dundi_context' is the name of the context being requested -; within the DUNDi request -; -; 'local_context' is the name of the context on the local system -; in which numbers can be looked up for which responses shall be given. -; -; 'weight' is the weight to use for the responses provided from this -; mapping. The number must be >= 0 and < 60000. Since it is totally -; valid to receive multiple responses to a query, responses received -; with a lower weight are tried first. Note that the weight has a -; special meaning in the e164 context - see the GPA for more details. -; -; 'tech' is the technology to use (IAX, SIP, H323) -; -; 'dest' is the destination to supply for reaching that number. The -; following variables can be used in the destination string and will -; be automatically substituted: -; ${NUMBER}: The number being requested -; ${IPADDR}: The IP address to connect to -; ${SECRET}: The current rotating secret key to be used -; -; Further options may include: -; -; nounsolicited: No unsolicited calls of any type permitted via this -; route -; nocomunsolicit: No commercial unsolicited calls permitted via -; this route -; residential: This number is known to be a residence -; commercial: This number is known to be a business -; mobile: This number is known to be a mobile phone -; nocomunsolicit: No commercial unsolicited calls permitted via -; this route -; nopartial: Do not search for partial matches -; -; There *must* exist an entry in mappings for DUNDi to respond -; to any request, although it may be empty. -; -;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial -;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial -;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial - -;digexten => default,0,IAX2,guest@lappy/${NUMBER} -;asdf => - -; -; Weights for mappings can be set a few different ways: -; -; 1) It can be set as a static number. -;testmap1 => context1,222,IAX2,guest@peer1/${NUMBER} -; -; 2) It can be an Asterisk global variable. -;testmap2 => context2,${DUNDITESTVAR},IAX2,guest@peer2${NUMBER} -; -; 3) It can be retrieved using a dialplan function. This can be extremely -; useful if you want to let an external script decide what the weight -; in a response shouuld be. -;testmap3 => context3,${SHELL(echo 123)},IAX2,guest@peer3/${NUMBER} -; -; The built in variables ${SECRET}, ${IPADDR} and ${NUMBER} can also be -; passed to the weight. For example, you could pass the ${NUMBER} value -; to your SHELL() script and use that to dynamically return a weight. -; -; Note than when using a global variable or dialplan function to set the -; weight for a mapping, that response caching should be disabled if you -; plan for these values to change frequently at all. If the results are -; cached, then any change in value will not take effect until the cache -; has expired. -; - -; -; The remaining sections represent the peers -; that we fundamentally trust. The section name -; represents the name and optionally at a specific -; DUNDi context if you want the trust to be established -; for only a specific DUNDi context. -; -; inkey - What key they will be authenticating to us with -; -; outkey - What key we use to authenticate to them -; -; host - What their host is -; -; port - The port where their host is listening (default: 4520) -; -; order - What search order to use. May be 'primary', 'secondary', -; 'tertiary' or 'quartiary'. In large systems, it is beneficial -; to only query one up-stream host in order to maximize caching -; value. Adding one with primary and one with secondary gives you -; redundancy without sacrificing performance. -; -; include - Includes this peer when searching a particular context -; for lookup (set "all" to perform all lookups with that -; host. This is also the context in which peers are permitted -; to precache. -; -; noinclude - Disincludes this peer when searching a particular context -; for lookup (set "all" to perform no lookups with that -; host. -; -; permit - Permits this peer to search a given DUNDi context on -; the local system. Set "all" to permit this host to -; lookup all contexts. This is also a context for which -; we will create/forward PRECACHE commands. -; -; deny - Denies this peer to search a given DUNDi context on -; the local system. Set "all" to deny this host to -; lookup all contexts. -; -; model - inbound, outbound, or symmetric for whether we receive -; requests only, transmit requests only, or do both. -; -; precache - Utilize/Permit precaching with this peer (to pre -; cache means to provide an answer when no request -; was made and is used so that machines with few -; routes can push those routes up a to a higher level). -; outgoing means we send precache routes to this peer, -; incoming means we permit this peer to send us -; precache routes. symmetric means we do both. -; -; Note: You cannot mix symmetric/outbound model with symmetric/inbound -; precache, nor can you mix symmetric/inbound model with symmetric/outbound -; precache. -; -; -; The '*' peer is special and matches an unspecified entity -; - -; -; Sample Primary e164 DUNDi peer -; -;[00:50:8B:F3:75:BB] -;model = symmetric -;host = 64.215.96.114 -;inkey = digium -;outkey = misery -;include = e164 -;permit = e164 -;qualify = yes - -; -; Sample Secondary e164 DUNDi peer -; -;[00:A0:C9:96:92:84] -;model = symmetric -;host = misery.digium.com -;inkey = misery -;outkey = ourkey -;include = e164 -;permit = e164 -;qualify = yes -;order = secondary - -; -; Sample "push mode" downstream host -; -;[00:0C:76:96:75:28] -;model = inbound -;host = dynamic -;precache = inbound -;inkey = littleguy -;outkey = ourkey -;include = e164 ; In this case used only for precaching -;permit = e164 -;qualify = yes - -; -; Sample "push mode" upstream host -; -;[00:07:E9:3B:76:60] -;model = outbound -;precache = outbound -;host = 216.207.245.34 -;register = yes -;inkey = dhcp34 -;permit = all ; In this case used only for precaching -;include = all -;qualify = yes -;outkey=foo - -;[*] -; diff --git a/version_1.9.1/pabx/etc_asterisk/enum.conf b/version_1.9.1/pabx/etc_asterisk/enum.conf deleted file mode 100644 index 39c7231..0000000 --- a/version_1.9.1/pabx/etc_asterisk/enum.conf +++ /dev/null @@ -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 diff --git a/version_1.9.1/pabx/etc_asterisk/extconfig.conf b/version_1.9.1/pabx/etc_asterisk/extconfig.conf deleted file mode 100644 index 9e13cac..0000000 --- a/version_1.9.1/pabx/etc_asterisk/extconfig.conf +++ /dev/null @@ -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]/ -; 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. diff --git a/version_1.9.1/pabx/etc_asterisk/extensions.ael b/version_1.9.1/pabx/etc_asterisk/extensions.ael deleted file mode 100644 index 495001f..0000000 --- a/version_1.9.1/pabx/etc_asterisk/extensions.ael +++ /dev/null @@ -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 -// -//