asterisk_user: asterisk
asterisk_group: asterisk

asterisk_dir: /var/lib/asterisk
asterisk_conf_dir: /etc/asterisk
asterisk_tls_dir: "{{ asterisk_conf_dir }}/tls"
asterisk_recordings_dir: /opt/recordings
asterisk_data_dir: "{{ asterisk_dir }}"

asterisk_users: {}
asterisk_trunks: {}

asterisk_language: ru

asterisk_pjsip_ciphers:
  - ECDHE-ECDSA-CHACHA20-POLY1305
  - ECDHE-ECDSA-AES256-GCM-SHA384
  - ECDHE-ECDSA-AES128-GCM-SHA256
  - ECDHE-RSA-CHACHA20-POLY1305
  - ECDHE-RSA-AES256-GCM-SHA384
  - ECDHE-RSA-AES128-GCM-SHA256
  - DHE-RSA-AES128-SHA256


# meta definitions:
# __template__ (bool) (section): this section is a template
# __template_from__ (string/list) (section): templates to inherit from
# __comment__ (string) (section): specify a comment before the section definition
# __inner_objects__ (boolean) (config/section): use object syntax when enumerating section members

asterisk_default_config:
  acl:
    acl_lan_clients:
      deny:
        - 0.0.0.0/0.0.0.0
      permit:
        - "{{ int_net | ansible.utils.ipaddr('network') }}/{{ int_net | ansible.utils.ipaddr('netmask') }}"
    acl_inet_clients:
      deny:
        - "{{ int_net | ansible.utils.ipaddr('network') }}/{{ int_net | ansible.utils.ipaddr('netmask') }}"
      permit:
        - 0.0.0.0/0.0.0.0

  asterisk:
    directories:
      __template__: yes
      __inner_objects__: yes
      astetcdir: "{{ asterisk_conf_dir }}"
      astvarlibdir: "{{ asterisk_dir }}"
      astdatadir: "{{ asterisk_data_dir }}"
      astdbdir: "{{ asterisk_db_dir | d(asterisk_dir) }}"
      astkeydir: "{{ asterisk_key_dir | d(asterisk_dir) }}"
      astagidir: "{{ asterisk_agi_dir | d(asterisk_dir ~ '/agi-bin') }}"
      astspooldir: "{{ asterisk_spool_dir | d('/var/spool/asterisk') }}"
      astrundir: "{{ asterisk_run_dir | d('/var/run/asterisk') }}"
      astlogdir: "{{ asterisk_log_dir | d('/var/log/asterisk') }}"
      astsbindir: /usr/sbin
      astmoddir: /usr/lib/asterisk/modules

    options:
      verbose: 0
      debug: no
      trace: 0

      execincludes: no
      highpriority: yes
      initcrypto: yes
      nocolor: yes
      dumpcore: no
      runuser: "{{ asterisk_user }}"
      rungroup: "{{ asterisk_group }}"
      autosystemname: yes
      maxcalls: 200
      maxload: "100.0"
      minmemfree: 1
      languageprefix: yes
      transmit_silence: no

      defaultlanguage: en
      documentation_language: en_US

  ccss:
    general:
      cc_max_requests: 15

  cdr:
    general:
      enable: yes
      unanswered: yes
      congestion: yes

  cel:
    general:
      enable: no

  cdr_pgsql:
    global:
      hostname: "{{ database_host }}"
      port: 5432
      user: "{{ database_user | d('cdr') }}"
      dbname: "{{ database_name | d('cdr') }}"
      table: "{{ database_table | d('cdr') }}"
      password: "{{ database_pass }}"
      encoding: UNICODE

  cli_aliases:
    general:
      template: friendly
    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_permissions:
    general:
      default_perm: permit

  codecs:
    plc:
      __inner_objects__: yes
      genericplc: "true"
      genericplc_on_equal_codecs: "false"
    opus:
      type: opus
      packet_loss: 2
      signal: voice

  confbridge:
    default_user:
      type: user
      dsp_drop_silence: yes
      jitterbuffer: yes
    default_bridge:
      type: bridge
      max_members: 30
      language: "{{ asterisk_language }}"

  features:
    __inner_objects__: yes
    featuremap:
      blindxfer: "**"
      atxfer: "*#"
    applicationmap:
      volume-up-tx: "#1,self/caller,Gosub(volume-up-tx,s,1)"
      volume-up-rx: "#2,self/caller,Gosub(volume-up-rx,s,1)"
      volume-down-tx: "#3,self/caller,Gosub(volume-down-tx,s,1)"
      volume-down-rx: "#4,self/caller,Gosub(volume-down-rx,s,1)"
      volume-increase-all: "#5,self/caller,Gosub(volume-increase-all,s,1)"
    call-controls:
      volume-up-tx: ""
      volume-up-rx: ""
      volume-down-tx: ""
      volume-down-rx: ""
      volume-increase-all: ""

  followme:
    __inner_objects__: yes
    general:
      featuredigittimeout: 3500
      enable_callee_prompt: "true"
      takecall: 1
      declinecall: 2
      call_from_prompt: followme/call-from
      norecording_prompt: followme/no-recording
      options_prompt: followme/options
      pls_hold_prompt: followme/pls-hold-while-try
      status_prompt: followme/status
      sorry_prompt: followme/sorry
      connecting_prompt: ""
    default:
      musicclass: default
      context: default
      enable_callee_prompt: "true"
      takecall: 1
      declinecall: 2
      call_from_prompt: followme/call-from
      norecording_prompt: followme/no-recording
      options_prompt: followme/options
      pls_hold_prompt: followme/pls-hold-while-try
      status_prompt: followme/status
      sorry_prompt: followme/sorry
      connecting_prompt: ""

  indications:
    general:
      country: ru
    ru:
      description: Russian Federation / ex Soviet Union
      ringcadence: "1000,4000"
      dial: "425"
      busy: "425/350,0/350"
      ring: "425/1000,0/4000"
      congestion: "425/175,0/175"
      callwaiting: "425/200,0/5000"
      record: "1400/400,0/15000"
      info: "950/330,1400/330,1800/330,0/1000"
      dialrecall: "425/400,0/40"
      stutter: "!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425"

  logger:
    general:
      queue_log: no
    logfiles:
      __inner_objects__: yes
      console: notice,warning,error,verbose,dtmf
      "syslog.local0": "[plain]notice,warning,error"

  manager:
    general:
      enabled: yes
      webenabled: no
      port: 5038
      bindaddr: 0.0.0.0
      debug: "off"
      allowmultiplelogin: yes
      displayconnects: yes
      timestampevents: yes
      authtimeout: 10

  musiconhold:
    default:
      mode: files
      directory: moh

  pjproject:
    startup:
      cache_pools: yes


  pjsip:
    system:
      type: system
      threadpool_auto_increment: 3
      timer_t1: 250
      timer_b: 16000
    global:
      type: global
      max_forwards: 40
      keep_alive_interval: 15
      user_agent: "{{ org }} Asterisk PBX"
      endpoint_identifier_order: username,ip
      default_from_user: pbx
      default_realm: "{{ host_fqdn }}"

    transport-common:
      __template__: yes
      type: transport
      tos: cs3
      cos: 3
      allow_reload: no
      local_net: "{{ int_net | ansible.utils.ipaddr('network') }}/{{ int_net | ansible.utils.ipaddr('netmask') }}"

    transport-ext:
      __template__: yes
      __template_from__: transport-common
      external_media_address: "{{ asterisk_external_ipv4 | d(hostvars[selected_node]['external_ipv4']) }}"
      external_signaling_address: "{{ asterisk_external_ipv4 | d(hostvars[selected_node]['external_ipv4']) }}"

    transport-udp:
      __template__: yes
      __template_from__: transport-common
      protocol: udp

    transport-tcp:
      __template__: yes
      __template_from__: transport-common
      protocol: tcp

    transport-lan:
      __template_from__: transport-udp
      bind: 0.0.0.0:5060

    transport-lan-tcp:
      __template_from__: transport-tcp
      bind: 0.0.0.0:5060

    transport-lan-tls:
      __template_from__: transport-common
      protocol: tls
      bind: 0.0.0.0:5061
      cert_file: "{{ asterisk_tls_dir }}/asterisk.crt"
      priv_key_file: "{{ asterisk_tls_dir }}/asterisk.key"
      cipher: "{{ asterisk_pjsip_ciphers | join(',') }}"
      method: tlsv1_2
      require_client_cert: no
      verify_client: no
      verify_server: no

    endpoint-common:
      __template__: yes
      type: endpoint
      allow: "!all,opus,g722,alaw,ulaw,g726,ilbc,gsm"
      allow_overlap: no
      send_connected_line: yes
      trust_connected_line: yes
      direct_media: no
      dtmf_mode: auto_info
      force_rport: yes
      ice_support: no
      identify_by: username
      rewrite_contact: yes
      rtp_symmetric: yes
      send_diversion: yes
      send_history_info: yes
      send_pai: no
      send_rpid: no
      use_ptime: yes
      t38_udptl: no
      tone_zone: ru
      language: ru
      tos_audio: ef
      cos_audio: 5
      rtp_keepalive: 5
      rtp_timeout: 360
      rtp_timeout_hold: 720
      rtcp_mux: yes
      max_video_streams: 0
      max_audio_streams: 1
      bundle: no
      sdp_session: "{{ org }} Asterisk PBX"
      sdp_owner: PBX
      suppress_q850_reason_headers: yes

    endpoint-trunk:
      __template__: yes
      __template_from__: endpoint-common
      identify_by: ip,username
      trust_id_inbound: yes
      acl: acl_inet_clients
      contact_acl: acl_inet_clients

    endpoint-lan:
      __template__: yes
      __template_from__: endpoint-common
      identify_by: username
      trust_id_inbound: no
      trust_id_outbound: yes
      acl: acl_lan_clients
      contact_acl: acl_lan_clients
      context: outbound
      allow_subscribe: yes
      device_state_busy_at: 1
      sub_min_expiry: 15
      media_encryption: sdes
      media_encryption_optimistic: yes

    auth-common:
      __template__: yes
      type: auth
      auth_type: userpass

    registration-common:
      __template__: yes
      type: registration
      expiration: 1800
      auth_rejection_permanent: no
      max_retries: 10000
      retry_interval: 20
      forbidden_retry_interval: 60
      fatal_retry_interval: 60

    aor-common:
      __template__: yes
      type: aor
      qualify_frequency: 30
      max_contacts: 2     # https://asterisk.org/pjsip-mis-configuration-can-cause-loss-sip-registrations

    __include__: custom_pjsip.conf


  pjsip_notify:
    __inner_objects__: yes
    clear-mwi:
      Event: message-summary
      Content-type: application/simple-message-summary
      Content:
        - "Messages-Waiting: no"
        - "Message-Account: sip:asterisk@127.0.0.1"
        - "Voice-Message: 0/0 (0/0)"
        - ""
    polycom-check-cfg:
      Event: check-sync
    yealink-reboot:
      Event: check-sync

  queues:
    general:
      persistentmembers: no
      autofill: yes
      monitor-type: MixMonitor
      updatecdr: yes
      log_membername_as_agent: yes
      shared_lastcall: yes

    queue-template:
      __template__: yes
      musicclass: default
      strategy: ringall
      servicelevel: 30
      maxlen: 128
      timeoutpriority: conf
      timeout: 300
      wrapuptime: 5
      announce-frequency: 0
      periodic-announce-frequency: 0
      announce-position: no
      autopause: yes
      autopausedelay: 60
      autopausebusy: yes
      joinempty: unavailable
      leavewhenempty: unavailable
      ringinuse: no

    queue-single:
      __template__: yes
      __template_from__: queue-template
      weight: 1
      autopause: no
      context: inbound-queued-inqueue-busy

    queue-le:
      __template__: yes
      __template_from__: queue-template
      weight: 1
      autopause: no

    __include__: custom_queues.conf

  queuerules:
    general:

  rtp:
    general:
      rtpstart: 15000
      rtpend: 19000
      strictrtp: yes
      icesupport: "false"

  udptl:
    general:

  modules:
    modules:
      autoload: no
      load:
        - app_attended_transfer.so
        - app_blind_transfer.so
        - app_bridgeaddchan.so
        - app_bridgewait.so
        - app_cdr.so
        - app_celgenuserevent.so
        - app_chanisavail.so
        - app_channelredirect.so
        - app_chanspy.so
        - app_confbridge.so
        - app_controlplayback.so
        - app_dial.so
        - app_directed_pickup.so
        - app_dumpchan.so
        - app_echo.so
        - app_exec.so
        - app_followme.so
        - app_forkcdr.so
        - app_mixmonitor.so
        - app_originate.so
        - app_playback.so
        - app_queue.so
        - app_read.so
        - app_readexten.so
        - app_senddtmf.so
        - app_softhangup.so
        - app_stack.so
        - app_stream_echo.so
        - app_talkdetect.so
        - app_transfer.so
        - app_verbose.so
        - app_waitforring.so
        - app_waitforsilence.so
        - app_waituntil.so
        - app_while.so

        - bridge_builtin_features.so
        - bridge_builtin_interval_features.so
        - bridge_holding.so
        - bridge_native_rtp.so
        - bridge_simple.so
        - bridge_softmix.so

        - cdr_pgsql.so

        - chan_bridge_media.so
        - chan_pjsip.so
        - chan_rtp.so

        - codec_a_mu.so
        - codec_adpcm.so
        - codec_alaw.so
        - codec_g722.so
        - codec_g726.so
        - codec_gsm.so
        - codec_ilbc.so
        - codec_opus_open_source.so
        - codec_resample.so
        - codec_ulaw.so

        - format_g719.so
        - format_g723.so
        - format_g726.so
        - format_gsm.so
        - format_ilbc.so
        - format_pcm.so
        - format_sln.so
        - format_vox.so
        - format_wav.so
        - format_wav_gsm.so

        - func_blacklist.so
        - func_callcompletion.so
        - func_callerid.so
        - func_cdr.so
        - func_channel.so
        - func_config.so
        - func_cut.so
        - func_devstate.so
        - func_dialplan.so
        - func_global.so
        - func_hangupcause.so
        - func_holdintercept.so
        - func_jitterbuffer.so
        - func_logic.so
        - func_module.so
        - func_pjsip_aor.so
        - func_pjsip_contact.so
        - func_pjsip_endpoint.so
        - func_rand.so
        - func_sorcery.so
        - func_strings.so
        - func_talkdetect.so
        - func_timeout.so
        - func_volume.so

        - pbx_config.so
        - pbx_loopback.so
        - pbx_realtime.so
        - pbx_spool.so

        - res_audiosocket.so
        - res_clialiases.so
        - res_clioriginate.so
        - res_convert.so
        - res_crypto.so
        - res_format_attr_celt.so
        - res_format_attr_g729.so
        - res_format_attr_ilbc.so
        - res_format_attr_opus.so
        - res_format_attr_silk.so
        - res_format_attr_siren14.so
        - res_format_attr_siren7.so
        - res_musiconhold.so
        - res_mutestream.so
        - res_pjproject.so

        - res_pjsip.so
        - res_pjsip_acl.so
        - res_pjsip_authenticator_digest.so
        - res_pjsip_caller_id.so
        - res_pjsip_dialog_info_body_generator.so
        - res_pjsip_diversion.so
        - res_pjsip_dlg_options.so
        - res_pjsip_dtmf_info.so
        - res_pjsip_empty_info.so
        - res_pjsip_endpoint_identifier_ip.so
        - res_pjsip_endpoint_identifier_user.so
        - res_pjsip_exten_state.so
        - res_pjsip_header_funcs.so
        - res_pjsip_history.so
        - res_pjsip_logger.so
        - res_pjsip_messaging.so
        - res_pjsip_mwi.so
        - res_pjsip_mwi_body_generator.so
        - res_pjsip_nat.so
        - res_pjsip_notify.so
        - res_pjsip_outbound_authenticator_digest.so
        - res_pjsip_outbound_publish.so
        - res_pjsip_outbound_registration.so
        - res_pjsip_path.so
        - res_pjsip_pidf_body_generator.so
        - res_pjsip_publish_asterisk.so
        - res_pjsip_pubsub.so
        - res_pjsip_refer.so
        - res_pjsip_registrar.so
        - res_pjsip_rfc3326.so
        - res_pjsip_sdp_rtp.so
        - res_pjsip_send_to_voicemail.so
        - res_pjsip_session.so
        - res_pjsip_sips_contact.so
        - res_pjsip_xpidf_body_generator.so

        - res_rtp_asterisk.so
        - res_rtp_multicast.so
        - res_security_log.so
        - res_sorcery_astdb.so
        - res_sorcery_config.so
        - res_sorcery_memory.so
        - res_sorcery_memory_cache.so
        - res_srtp.so
        - res_stasis.so
        - res_stasis_answer.so
        - res_stasis_device_state.so
        - res_stasis_playback.so
        - res_stasis_recording.so
        - res_timing_pthread.so
        - res_timing_timerfd.so

        - res_pjsip_header_funcs.so
        - res_pjsip_history.so
        - res_pjsip_sdp_rtp.so