242 lines
10 KiB
XML
242 lines
10 KiB
XML
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<include>
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<!-- Preprocessor Variables
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These are introduced when configuration strings must be consistent across modules.
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NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
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WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
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YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
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toll fraud in the future. It's your responsibility to secure your own system.
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This default config is used to demonstrate the feature set of FreeSWITCH.
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WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
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-->
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<X-PRE-PROCESS cmd="set" data="default_password=1234"/>
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<!-- Did you change it yet? -->
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<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
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<!--
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This setting is what sets the default domain FreeSWITCH will use if all else fails.
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FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does
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affect the sip authentication. Please review conf/directory/default.xml for more
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information on this topic.
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-->
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<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
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<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
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<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
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<X-PRE-PROCESS cmd="set" data="use_profile=internal"/>
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<!--
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Enable ZRTP globally you can override this on a per channel basis
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http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
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-->
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<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
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<!--
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Examples of codec options: (module must be compiled and loaded)
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codecname[@8000h|16000h|32000h[@XXi]]
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XX is the frame size must be multples allowed for the codec
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FreeSWITCH can support 10-120ms on some codecs.
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We do not support exceeding the MTU of the RTP packet.
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iLBC@30i - iLBC using mode=30 which will win in all cases.
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DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
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DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
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speex@8000h@20i - Speex 8kHz using 20ms ptime.
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speex@16000h@20i - Speex 16kHz using 20ms ptime.
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speex@32000h@20i - Speex 32kHz using 20ms ptime.
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BV16 - BroadVoice 16kb/s narrowband, 8kHz
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BV32 - BroadVoice 32kb/s wideband, 16kHz
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G7221@16000h - G722.1 16kHz (aka Siren 7)
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G7221@32000h - G722.1C 32kHz (aka Siren 14)
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CELT@32000h - CELT 32kHz, only 10ms supported
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CELT@48000h - CELT 48kHz, only 10ms supported
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GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
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G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
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PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
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PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
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G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
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AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
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AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
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AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
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AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
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LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
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L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
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These are the passthru audio codecs:
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G729 - G729 in passthru mode. (mod_g729)
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G723 - G723.1 in passthru mode. (mod_g723_1)
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AMR - AMR in passthru mode. (mod_amr)
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These are the passthru video codecs: (mod_h26x)
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H261 - H.261 Video
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H263 - H.263 Video
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H263-1998 - H.263-1998 Video
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H263-2000 - H.263-2000 Video
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H264 - H.264 Video
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RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.
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96 - AMR
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97 - iLBC (30)
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98 - iLBC (20)
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99 - Speex 8kHz, 16kHz, 32kHz
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100 -
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101 - telephone-event
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102 -
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103 -
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104 -
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105 -
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106 - BV16
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107 - G722.1 (16kHz)
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108 -
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109 -
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110 -
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111 -
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112 -
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113 -
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114 - CELT 32kHz, 48kHz
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115 - G722.1C (32kHz)
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116 -
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117 - SILK 8kHz
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118 - SILK 12kHz
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119 - SILK 16kHz
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120 - SILK 24kHz
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121 - AAL2-G726-40 && G726-40
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122 - AAL2-G726-32 && G726-32
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123 - AAL2-G726-24 && G726-24
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124 - AAL2-G726-16 && G726-16
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125 -
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126 -
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127 - BV32
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-->
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<X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20iG7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
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<!--
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xmpp_client_profile and xmpp_server_profile
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xmpp_client_profile can be any string.
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xmpp_server_profile is appended to "dingaling_" to form the database name
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containing the "subscriptions" table.
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used by: dingaling.conf.xml enum.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
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<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
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<!--
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THIS IS ONLY USED FOR DINGALING
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bind_server_ip
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Can be an ip address, a dns name, or "auto".
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This determines an ip address available on this host to bind.
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If you are separating RTP and SIP traffic, you will want to have
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use different addresses where this variable appears.
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Used by: dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
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<!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
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If you're going to load test FreeSWITCH please input real IP addresses
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for external_rtp_ip and external_sip_ip
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-->
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<!-- external_rtp_ip
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Can be an one of:
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ip address: "12.34.56.78"
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a stun server lookup: "stun:stun.server.com"
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a DNS name: "host:host.server.com"
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where fs.mydomain.com is a DNS A record-useful when fs is on
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a dynamic IP address, and uses a dynamic DNS updater.
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If unspecified, the bind_server_ip value is used.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
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<!-- external_sip_ip
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Used as the public IP address for SDP.
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Can be an one of:
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ip address: "12.34.56.78"
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a stun server lookup: "stun:stun.server.com"
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a DNS name: "host:host.server.com"
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where fs.mydomain.com is a DNS A record-useful when fs is on
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a dynamic IP address, and uses a dynamic DNS updater.
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If unspecified, the bind_server_ip value is used.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
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<!-- unroll-loops
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Used to turn on sip loopback unrolling.
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-->
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<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
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<!-- outbound_caller_id and outbound_caller_name
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The caller ID telephone number we should use when calling out.
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Used by: conference.conf.xml and user directory for default
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outbound callerid name and number.
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-->
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<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
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<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
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<!-- various debug and defaults -->
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<X-PRE-PROCESS cmd="set" data="call_debug=false"/>
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<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
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<X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
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<X-PRE-PROCESS cmd="set" data="default_country=US"/>
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<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
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<X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440.0,480.0)"/>
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<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440.0,0.0)"/>
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<X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425.0,0.0)"/>
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<X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425,0)"/>
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<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
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<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
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<!--
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Setting up your default sip provider is easy.
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Below are some values that should work in most cases.
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These are for conf/directory/default/example.com.xml
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-->
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<X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
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<X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
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<X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
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<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
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<!-- true or false -->
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<X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
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<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
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<!--
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SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
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-->
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<X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>
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<!-- Internal SIP Profile -->
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<X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
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<X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
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<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
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<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
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<X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
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<!-- External SIP Profile -->
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<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
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<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
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<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
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<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
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<X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>
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</include>
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