freeswitch/conf/vars.xml

  <!-- Preprocessor Variables
       These are introduced when configuration strings must be consistent across modules. 
   -->
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="moh_uri=local_stream://moh"/>
  <!-- sip_profile 
       Must be a domain name if you are being a registry server; otherwise
       can be any string.
       used by: sofia.conf.xml enum.conf.xml default_context.xml directory.xml
  --> 
  <X-PRE-PROCESS cmd="set" data="sip_profile=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="nat_sip_profile=nat_$${domain}"/>
  <!-- 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 
  --> 
  <!-- 
       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.
       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

  -->
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU@20i,PCMA@20i,H263"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i"/>
  <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
  <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
  <!-- 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: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
  <!-- external_rtp_ip
       Used as the public IP address for SDP.
       Can be an ip address or a string like "stun:stun.server.com" 
       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 ip address or a string like "stun:stun.server.com" 
       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
  -->
  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=8777423583"/>
  <X-PRE-PROCESS cmd="set" data="call_debug=false"/>