Config vars.xml

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  • outbound_caller_name - Name shown if phones when you make outbound calls
  • outbound_caller_number - Number shown in phones when you make outbound calls
  • domain - Domain that users register to (domain is specified in phone configs).
  • external_rtp_ip and external_sip_ip - Safest to use an IP address rather than DNS, in case a device can't resolve the domain.
  • NOTE: The example below may not be the most up-to-date default version that you will find in your installation.


Related info from the FreeSwitch FAQ:

Q: What is the difference between using a ${var} and $${var} in the configuration files?

A: The ${var} is expanded as its encountered in the dialplan. The $${var} variation is used as a preprocessor variable and is expanded at load or reloadxml. See vars.xml in the conf folder for more information.

Predefined variables

The following variables are set dynamically - claculated 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
core_uuid
zrtp_enabled
nat_public_addr
nat_private_addr
nat_type

In my tests running with -nonat the nat_xxx ones above do not exist

  <!-- 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? -->

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

  <!--
      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=internal"/>

  <!--
      Enable ZRTP globally you can override this on a per channel basis
      
      http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
  -->
  <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>

  <!-- 
       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 what for.

       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=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

  <!--
      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="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="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"/>

  <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
  <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440.0,480.0)"/>
  <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440.0,0.0)"/>
  <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425.0,0.0)"/>
  <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425,0)"/>
  <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="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
  <!--
      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
  -->
  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>

  <!-- 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"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- 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"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

See Also: