WhitespaceClean-resources/templates/conf
whitespace pass over files for reference regex that was used s/[ \t]+(\r?\n)/\1/
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@@ -1,6 +1,6 @@
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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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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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-->
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@@ -9,8 +9,8 @@
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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 $${local_ip_v4} unless changed. Changing this setting does
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FreeSWICH will default to $${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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@@ -21,18 +21,18 @@
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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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<!--
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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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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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@@ -62,21 +62,21 @@
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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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@@ -86,9 +86,9 @@
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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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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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@@ -108,7 +108,7 @@
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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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125 -
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126 -
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127 - BV32
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@@ -118,20 +118,20 @@
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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_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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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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<!--
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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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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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@@ -140,7 +140,7 @@
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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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@@ -172,7 +172,7 @@
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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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-->
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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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@@ -226,7 +226,7 @@
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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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