freeswitch/conf/vanilla/vars.xml

<include>
  <!-- Preprocessor Variables
       These are introduced when configuration strings must be consistent across modules.
       NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.

       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING

       YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
       toll fraud in the future.  It's your responsibility to secure your own system.

       This default config is used to demonstrate the feature set of FreeSWITCH.

       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
  -->
  <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->
  <!--
      The following variables are set dynamically - calculated if possible by freeswitch - and
      are available to the config as $${variable}.  You can see their calculated value via fs_cli
      by entering eval $${variable}

      hostname
      local_ip_v4
      local_mask_v4
      local_ip_v6
      switch_serial
      base_dir
      recordings_dir
      sound_prefix
      sounds_dir
      conf_dir
      log_dir
      run_dir
      db_dir
      mod_dir
      htdocs_dir
      script_dir
      temp_dir
      grammar_dir
      certs_dir
      storage_dir
      cache_dir
      core_uuid
      nat_public_addr
      nat_private_addr
      nat_type

  -->


  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
  <!--<Z-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/allison"/> -->

  <!--
      This setting is what sets the default domain FreeSWITCH will use if all else fails.

      FreeSWICH will default to $${local_ip_v4} unless changed.  Changing this setting does
      affect the sip authentication.  Please review conf/directory/default.xml for more
      information on this topic.
  -->
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="use_profile=external"/>
  <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
  <!--
      NOTICE: When using SRTP it's critical that you do not offer or accept
      variable bit rate codecs, doing so would leak information and possibly
      compromise your SRTP stream. (FS-6404)

      Supported SRTP Crypto Suites:

      AEAD_AES_256_GCM_8
      ____________________________________________________________________________
      This algorithm is identical to AEAD_AES_256_GCM (see Section 5.2 of
      [RFC5116]), except that the tag length, t, is 8, and an
      authentication tag with a length of 8 octets (64 bits) is used.
      An AEAD_AES_256_GCM_8 ciphertext is exactly 8 octets longer than its
      corresponding plaintext.


      AEAD_AES_128_GCM_8
      ____________________________________________________________________________
      This algorithm is identical to AEAD_AES_128_GCM (see Section 5.1 of
      [RFC5116]), except that the tag length, t, is 8, and an
      authentication tag with a length of 8 octets (64 bits) is used.
      An AEAD_AES_128_GCM_8 ciphertext is exactly 8 octets longer than its
      corresponding plaintext.


      AES_CM_256_HMAC_SHA1_80 | AES_CM_192_HMAC_SHA1_80 | AES_CM_128_HMAC_SHA1_80
      ____________________________________________________________________________
      AES_CM_128_HMAC_SHA1_80 is the SRTP default AES Counter Mode cipher
      and HMAC-SHA1 message authentication with an 80-bit authentication
      tag. The master-key length is 128 bits and has a default lifetime of
      a maximum of 2^48 SRTP packets or 2^31 SRTCP packets, whichever comes
      first.


      AES_CM_256_HMAC_SHA1_32 | AES_CM_192_HMAC_SHA1_32 | AES_CM_128_HMAC_SHA1_32
      ____________________________________________________________________________
      This crypto-suite is identical to AES_CM_128_HMAC_SHA1_80 except that
      the authentication tag is 32 bits. The length of the base64-decoded key and
      salt value for this crypto-suite MUST be 30 octets i.e., 240 bits; otherwise,
      the crypto attribute is considered invalid.


      AES_CM_128_NULL_AUTH
      ____________________________________________________________________________
      The SRTP default cipher (AES-128 Counter Mode), but to use no authentication
      method.  This policy is NOT RECOMMENDED unless it is unavoidable; see
      Section 7.5 of [RFC3711].


      SRTP variables that modify behaviors based on direction/leg:

      rtp_secure_media
      ____________________________________________________________________________
      possible values:
          mandatory - Accept/Offer SAVP negotiation ONLY
          optional  - Accept/Offer SAVP/AVP with SAVP preferred
          forbidden - More useful for inbound to deny SAVP negotiation
          false     - implies forbidden
          true      - implies mandatory

      default if not set is accept SAVP inbound if offered.


      rtp_secure_media_inbound | rtp_secure_media_outbound
      ____________________________________________________________________________
      This is the same as rtp_secure_media, but would apply to either inbound
      or outbound offers specifically.


      How to specify crypto suites:
      ____________________________________________________________________________
      By default without specifying any crypto suites FreeSWITCH will offer
      crypto suites from strongest to weakest accepting the strongest each
      endpoint has in common.  If you wish to force specific crypto suites you
      can do so by appending the suites in a colon separated list in the order
      that you wish to offer them in.

      Examples:

          rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80:AES_CM_256_HMAC_SHA1_32
          rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80:AES_CM_256_HMAC_SHA1_32
          rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
          rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80

      Additionally you can narrow this down on either inbound or outbound by
      specifying as so:

          rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8
          rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8
          rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8
          rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8


      rtp_secure_media_suites
      ____________________________________________________________________________
      Optionaly you can use rtp_secure_media_suites to dictate the suite list
      and only use rtp_secure_media=[optional|mandatory|false|true] without having
      to dictate the suite list with the rtp_secure_media* variables.
  -->
  <!--
       Examples of codec options: (module must be compiled and loaded)

       codecname[@8000h|16000h|32000h[@XXi]]

       XX is the frame size must be multples allowed for the codec
       FreeSWITCH can support 10-120ms on some codecs.
       We do not support exceeding the MTU of the RTP packet.


       iLBC@30i         - iLBC using mode=30 which will win in all cases.
       DVI4@8000h@20i   - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
       DVI4@16000h@40i  - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
       speex@8000h@20i  - Speex 8kHz using 20ms ptime.
       speex@16000h@20i - Speex 16kHz using 20ms ptime.
       speex@32000h@20i - Speex 32kHz using 20ms ptime.
       BV16             - BroadVoice 16kb/s narrowband, 8kHz
       BV32             - BroadVoice 32kb/s wideband, 16kHz
       G7221@16000h     - G722.1 16kHz (aka Siren 7)
       G7221@32000h     - G722.1C 32kHz (aka Siren 14)
       CELT@32000h      - CELT 32kHz, only 10ms supported
       CELT@48000h      - CELT 48kHz, only 10ms supported
       GSM@40i          - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
       G722             - G722 16kHz using default 20ms ptime. (multiples of 10)
       PCMU             - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
       PCMA             - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
       G726-16          - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-24          - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-32          - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-40          - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
       AAL2-G726-16     - Same as G726-16 but using AAL2 packing. (multiples of 10)
       AAL2-G726-24     - Same as G726-24 but using AAL2 packing. (multiples of 10)
       AAL2-G726-32     - Same as G726-32 but using AAL2 packing. (multiples of 10)
       AAL2-G726-40     - Same as G726-40 but using AAL2 packing. (multiples of 10)
       LPC              - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
       L16              - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.

       These are the passthru audio codecs:

       G729             - G729 in passthru mode. (mod_g729)
       G723             - G723.1 in passthru mode. (mod_g723_1)
       AMR              - AMR in passthru mode. (mod_amr)

       These are the passthru video codecs: (mod_h26x)

       H261             - H.261 Video
       H263             - H.263 Video
       H263-1998        - H.263-1998 Video
       H263-2000        - H.263-2000 Video
       H264             - H.264 Video

       RTP Dynamic Payload Numbers currently used in FreeSWITCH and their purpose.

       96  - AMR
       97  - iLBC (30)
       98  - iLBC (20)
       99  - Speex 8kHz, 16kHz, 32kHz
       100 -
       101 - telephone-event
       102 -
       103 -
       104 -
       105 -
       106 - BV16
       107 - G722.1 (16kHz)
       108 -
       109 -
       110 -
       111 -
       112 -
       113 -
       114 - CELT 32kHz, 48kHz
       115 - G722.1C (32kHz)
       116 -
       117 - SILK 8kHz
       118 - SILK 12kHz
       119 - SILK 16kHz
       120 - SILK 24kHz
       121 - AAL2-G726-40 && G726-40
       122 - AAL2-G726-32 && G726-32
       123 - AAL2-G726-24 && G726-24
       124 - AAL2-G726-16 && G726-16
       125 -
       126 -
       127 - BV32

  -->
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>

  <!--
      xmpp_client_profile and xmpp_server_profile
      xmpp_client_profile can be any string.
      xmpp_server_profile is appended to "dingaling_" to form the database name
      containing the "subscriptions" table.
      used by: dingaling.conf.xml enum.conf.xml
  -->

  <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
  <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
  <!--
       THIS IS ONLY USED FOR DINGALING

       bind_server_ip

       Can be an ip address, a dns name, or "auto".
       This determines an ip address available on this host to bind.
       If you are separating RTP and SIP traffic, you will want to have
       use different addresses where this variable appears.
       Used by: dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

  <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE

       If you're going to load test FreeSWITCH please input real IP addresses
       for external_rtp_ip and external_sip_ip
  -->

  <!-- external_rtp_ip
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

  <!-- external_sip_ip
      Used as the public IP address for SDP.
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/>

  <!-- unroll-loops
       Used to turn on sip loopback unrolling.
  -->
  <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>

  <!-- outbound_caller_id and outbound_caller_name
       The caller ID telephone number we should use when calling out.
       Used by: conference.conf.xml and user directory for default
       outbound callerid name and number.
  -->
  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

  <!-- various debug and defaults -->
  <X-PRE-PROCESS cmd="set" data="call_debug=false"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
  <X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
  <X-PRE-PROCESS cmd="set" data="default_country=US"/>

  <!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
       if true, the destination number is not included -->
  <X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>

  <X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/>
  <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
  <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
  <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
  <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
  <X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
  <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
  <X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
  <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
  <X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
  <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
  <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
  <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
  <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
  <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
  <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
  <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
  <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
  <X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/>
  <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>

  <!--
       Digits Dialed filter: (FS-6940)

       The digits stream may contain valid credit card numbers or social security numbers, These digit
       filters will allow you to make a valant effort to stamp out sensitive information for
       PCI/HIPPA compliance. (see xml_cdr dialed_digits)

       df_us_ssn   = US Social Security Number pattern
       df_us_luhn  = Visa, MasterCard, American Express, Diners Club, Discover and JCB
  -->
  <X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/>
  <X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/>
  <!-- change XX to X below to enable -->
  <XX-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/>

  <!--
      Setting up your default sip provider is easy.
      Below are some values that should work in most cases.

      These are for conf/directory/default/example.com.xml
  -->
  <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
  <!-- true or false -->
  <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>

  <!--
     SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls

     valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2

     default: tlsv1,tlsv1.1,tlsv1.2
  -->
  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>

  <!--
     TLS cipher suite: default ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH

     The actual ciphers supported will change per platform.

     openssl ciphers -v 'ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH'

     Will show you what is available in your verion of openssl.
  -->
  <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>

  <!-- Video Settings -->
  <!-- Setting the max bandwdith -->
  <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=3mb"/>
  <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=3mb"/>

  <!-- WebRTC Video -->
  <!-- Suppress CNG for WebRTC Audio -->
  <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
  <!-- Enable liberal DTMF for those that can't get it right -->
  <X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
  <!-- Helps with WebRTC Audio -->

  <!-- Stock Video Avatars -->
  <X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/>
  <X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/>

</include>