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      <name>Freeswitch</name>
      <created_at>2006-12-21T01:13:50Z</created_at>
      <updated_at>2009-11-24T11:34:29Z</updated_at>
      <description>FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch.  It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.</description>
      <homepage_url>http://www.freeswitch.org/</homepage_url>
      <download_url>http://files.freeswitch.org/</download_url>
      <url_name></url_name>
      <user_count>11</user_count>
      <average_rating></average_rating>
      <rating_count>3</rating_count>
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        <project_id>3853</project_id>
        <updated_at>2009-01-23T15:34:20Z</updated_at>
        <logged_at>2008-09-05T19:29:18Z</logged_at>
        <min_month>2007-02-01T00:00:00Z</min_month>
        <max_month>2008-09-01T00:00:00Z</max_month>
        <twelve_month_contributor_count>5</twelve_month_contributor_count>
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        <main_language_name>C</main_language_name>
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      <licenses>
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          <name>mozilla_public_1_1</name>
          <nice_name>Mozilla Public License 1.1</nice_name>
        </license>
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