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add dialing by sip (thanks trixter)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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@ -36,6 +36,13 @@
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</condition>
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</condition>
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</extension>
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</extension>
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<!-- dial via SIP uri -->
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<extension name="SIP_URI">
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<condition field="destination_number" expression="^sip:(.*)$">
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<action application="bridge" data="sofia/${use_profile}/$1"/>
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</condition>
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</extension>
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<!--
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<!--
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start a dynamic conference with the settings of the
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start a dynamic conference with the settings of the
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"default" conference profile in conference.conf.xml
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"default" conference profile in conference.conf.xml
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