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more tweaks and docs
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9715 d0543943-73ff-0310-b7d9-9358b9ac24b2
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Brian West
parent
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@ -1,3 +1,14 @@
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<!--
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NOTICE:
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This context is usually accessed via authenticated or transfered callers from the public context
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from the sip profile sitting on port 5060.
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Authenitcated users will use the user_context variable on the user to determin what context
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they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml
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will build the domains acl using this value.
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-->
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<?xml version="1.0" encoding="utf-8"?>
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<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
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<include>
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@ -82,7 +93,7 @@
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<!-- If to_host and from_host differ this is an external sip URI so lets send it out. -->
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<extension name="external_sip_uri" continue="true">
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<condition field="source" expression="mod_sofia"/>
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<condition field="${outside_call}" expression="^$"/>
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<condition field="${outside_call}"a expression="^$"/>
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<condition field="${sip_req_host}" expression="${sip_from_host}">
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<anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/>
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</condition>
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@ -106,9 +117,6 @@
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</condition>
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</extension>
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<extension name="eavesdrop">
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<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
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<action application="answer"/>
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@ -350,14 +358,14 @@
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is to use phone's star codes (defaults to *38 and *39).
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-->
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<extension name="park">
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<condition field="source" expression="mod_sofia"/>
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<condition field="source" expression="mod_sofia"/>
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<condition field="destination_number" expression="callpark"/>
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<condition field="${sip_refer_to}">
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<expression><![CDATA[<sip:callpark@${domain};orbit=(\d+)>]]></expression>
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<action application="fifo" data="$1@${domain} in undef $${hold_music}"/>
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</condition>
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<condition field="${sip_refer_to}">
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<expression><![CDATA[<sip:callpark@${domain};orbit=(\d+)>]]></expression>
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<action application="fifo" data="$1@${domain} in undef $${hold_music}"/>
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</condition>
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</extension>
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<!--
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This extension is used with linksys phones.
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@ -366,11 +374,11 @@
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and we extract that info.
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-->
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<extension name="unpark">
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<condition field="source" expression="mod_sofia"/>
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<condition field="source" expression="mod_sofia"/>
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<condition field="destination_number" expression="pickup"/>
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<condition field="${sip_to_params}" expression="orbit\=(\d+)">
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<action application="answer"/>
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<action application="fifo" data="$1@${domain} out nowait"/>
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<condition field="${sip_to_params}" expression="orbit\=(\d+)">
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<action application="answer"/>
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<action application="fifo" data="$1@${domain} out nowait"/>
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</condition>
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</extension>
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@ -428,6 +436,11 @@
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</condition>
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</extension>
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<!--
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You will no longer hear the bong tone. The wav file is playing stating the call is secure.
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The file will not play unless you have both TLS and SRTP active.
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-->
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<extension name="hold_music">
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<condition field="destination_number" expression="^9999$"/>
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<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
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@ -440,7 +453,23 @@
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</condition>
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</extension>
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<!--
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You can place files in the default directory to get included.
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-->
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<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
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<!--
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WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
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Anything you put below this line will usually get ignored to to the file in
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default/99999_enum.xml as it will transfer the call to the enum dialplan.
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WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
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-->
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<!--
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This is an example of how to overide the RURI on an outgoing invite to a registered contact.
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-->
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<!--
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<extension name="refer">
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<condition field="${sip_refer_to}">
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@ -454,11 +483,7 @@
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<action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/>
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</condition>
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</extension>
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-->
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<!--
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This is an example of how to overide the RURI on an outgoing invite to a registered contact.
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-->
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<!--
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<extension name="ruri">
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<condition field="destination_number" expression="^ruri$">
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<action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
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@ -475,5 +500,7 @@
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</extension>
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-->
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<!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! -->
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</context>
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</include>
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@ -1,13 +1,18 @@
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<!--
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NOTICE:
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This context is usually accessed via the external sip profile sitting on port 5080.
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It is recommended to have separate inbound and outbound contexts. Not only for security
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but clearing up why you would need to do such a thing. You don't want outside un-authenticated
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callers hitting your default context which allows dialing calls thru your providers and results
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in Toll Fraud.
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-->
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<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
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<include>
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<context name="public">
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<extension name="outside_call" continue="true">
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<condition break="never">
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<action application="set" data="outside_call=true"/>
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</condition>
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</extension>
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<extension name="unloop">
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<condition field="${unroll_loops}" expression="^true$"/>
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<condition field="${sip_looped_call}" expression="^true$">
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</condition>
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</extension>
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<!--
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Tag anything pass thru here as an outside_call so you can make sure not
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to create any routing loops based on the conditions that it came from
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the outside of the switch.
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-->
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<extension name="outside_call" continue="true">
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<condition break="never">
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<action application="set" data="outside_call=true"/>
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</condition>
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</extension>
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<!-- Try to get target_domain from the sip_req_params variable -->
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<extension name="set_domain" continue="true">
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<condition field="${target_domain}" expression="^$" break="never"/>
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@ -37,7 +53,10 @@
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<action application="transfer" data="$1 XML default"/>
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</condition>
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</extension>
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<!--
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You can place files in the public directory to get included.
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-->
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<X-PRE-PROCESS cmd="include" data="public/*.xml"/>
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</context>
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