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    <stack>
      <id>30353</id>
      <title>VOIPPIX</title>
      <description>VOIP SIP software</description>
      <updated_at>2009-04-23T15:58:00Z</updated_at>
      <project_count>19</project_count>
      <stack_entries>
        <stack_entry>
          <id>125196</id>
          <stack_id>30353</stack_id>
          <project_id>7285</project_id>
          <created_at>2008-05-07T08:28:39Z</created_at>
          <project>
            <id>7285</id>
            <name>Ekiga</name>
            <created_at>2007-08-08T04:11:16Z</created_at>
            <updated_at>2009-12-26T04:18:15Z</updated_at>
            <description>Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting.</description>
            <homepage_url>http://ekiga.org/</homepage_url>
            <download_url>http://ekiga.org/?rub=5</download_url>
            <url_name>ekiga</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/1308/ekiga_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/1308/ekiga_small.png</small_logo_url>
            <user_count>58</user_count>
            <average_rating>3.92857</average_rating>
            <rating_count>14</rating_count>
            <analysis_id>810078</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>125197</id>
          <stack_id>30353</stack_id>
          <project_id>63</project_id>
          <created_at>2008-05-07T08:28:54Z</created_at>
          <project>
            <id>63</id>
            <name>Pidgin</name>
            <created_at>2006-10-10T15:51:31Z</created_at>
            <updated_at>2009-11-28T23:28:29Z</updated_at>
            <description>Pidgin is an instant messaging program for Windows, Linux, BSD, and other Unixes. You can talk to your friends using AIM, ICQ, Jabber/XMPP, MSN Messenger, Yahoo!, Bonjour, Gadu-Gadu, IRC, Novell GroupWise Messenger, QQ, Lotus Sametime, SILC, SIMPLE, and Zephyr.

Pidgin can log in to multiple accounts on multiple IM networks simultaneously. This means that you can be chatting with friends on AIM, talking to a friend on Yahoo Messenger, and sitting in an IRC channel all at the same time.

Pidgin supports many features of the various networks, such as file transfer, away messages, and typing notification. It also goes beyond that and provides many unique features.

[Previously known as &quot;Gaim&quot;.]</description>
            <homepage_url>http://www.pidgin.im/</homepage_url>
            <download_url>http://pidgin.im/download/</download_url>
            <url_name>pidgin</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/1340/Pidgin_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/1340/Pidgin_small.png</small_logo_url>
            <user_count>1426</user_count>
            <average_rating>4.11916</average_rating>
            <rating_count>427</rating_count>
            <analysis_id>629991</analysis_id>
            <licenses>
              <license>
                <name>gpl2_or_later</name>
                <nice_name>GNU General Public License 2.0 or later</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>125198</id>
          <stack_id>30353</stack_id>
          <project_id>96</project_id>
          <created_at>2008-05-07T08:31:20Z</created_at>
          <project>
            <id>96</id>
            <name>Asterisk</name>
            <created_at>2006-10-10T15:51:31Z</created_at>
            <updated_at>2009-12-27T04:29:49Z</updated_at>
            <description>Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris.  It provides all of the features you would expect from a PBX and more as it enables developers to build customized voice applications of many types.  Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.</description>
            <homepage_url>http://www.asterisk.org/</homepage_url>
            <download_url>http://www.asterisk.org/downloads/</download_url>
            <url_name>asterisk</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/2352/asterisk_med.gif</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/2352/asterisk_small.gif</small_logo_url>
            <user_count>118</user_count>
            <average_rating>4.34286</average_rating>
            <rating_count>35</rating_count>
            <analysis_id>811218</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>125199</id>
          <stack_id>30353</stack_id>
          <project_id>10686</project_id>
          <created_at>2008-05-07T08:31:57Z</created_at>
          <project>
            <id>10686</id>
            <name>FreePBX</name>
            <created_at>2008-01-03T08:29:05Z</created_at>
            <updated_at>2009-12-26T21:02:45Z</updated_at>
            <description>FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any &quot;built in&quot; programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. 

Some of the features include voicemail, IVR menus, conferencing, paging, ring groups, call routing, queues, and many more.</description>
            <homepage_url>http://freepbx.org</homepage_url>
            <download_url>http://freepbx.org/download-freepbx</download_url>
            <url_name>amportal</url_name>
            <user_count>14</user_count>
            <average_rating>4.66667</average_rating>
            <rating_count>6</rating_count>
            <analysis_id>810767</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>125200</id>
          <stack_id>30353</stack_id>
          <project_id>3657</project_id>
          <created_at>2008-05-07T08:32:13Z</created_at>
          <project>
            <id>3657</id>
            <name>OpenPBX</name>
            <created_at>2006-11-23T01:53:11Z</created_at>
            <updated_at>2008-05-07T08:32:14Z</updated_at>
            <description>OpenPBX.org is pleased to announce that the OpenPBX.org software PBX is now running on multiple operating systems.  On November 22, 2006 calls were passed using MacOS X on the PowerPC architecture. Calls have also been completed using NetBSD, FreeBSD 6.2 and 7. These are first steps in supporting the goal of running OpenPBX.org on multiple operating systems and computer architectures.</description>
            <homepage_url>http://www.openpbx.org/</homepage_url>
            <download_url></download_url>
            <url_name></url_name>
            <user_count>3</user_count>
            <average_rating></average_rating>
            <rating_count>0</rating_count>
            <analysis_id>112670</analysis_id>
            <licenses>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>240604</id>
          <stack_id>30353</stack_id>
          <project_id>318963</project_id>
          <created_at>2009-04-23T15:56:10Z</created_at>
          <project>
            <id>318963</id>
            <name>SFLphone</name>
            <created_at>2009-03-11T14:44:35Z</created_at>
            <updated_at>2009-12-22T20:11:04Z</updated_at>
            <description>SFLphone is a SIP/IAX2 compatible softphone for Linux. The SFLphone project's goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed with a hundred-calls-a-day receptionist in mind.</description>
            <homepage_url>http://www.sflphone.org/</homepage_url>
            <download_url>http://www.sflphone.org/download.php</download_url>
            <url_name>sflphone</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/27876/logo-bg_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/27876/logo-bg_small.png</small_logo_url>
            <user_count>2</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>804894</analysis_id>
            <licenses>
              <license>
                <name>gpl3</name>
                <nice_name>GNU General Public License 3</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>240606</id>
          <stack_id>30353</stack_id>
          <project_id>7704</project_id>
          <created_at>2009-04-23T15:58:00Z</created_at>
          <project>
            <id>7704</id>
            <name>Zoiper Free IAX and SIP softphone</name>
            <created_at>2007-08-13T15:56:53Z</created_at>
            <updated_at>2009-02-25T17:12:57Z</updated_at>
            <description>ZOIPER 2.0 Free SIP and IAX is a user-friendly softphone, compatible with Asterisk or any other SIP or IAX capable system. Great VoIP calling experience with features like: 

SIP + IAX / IAX 2 protocols
Available codecs &#8211; GSM, ulaw, alaw, speex, ilbc
STUN support
STUN server per account
Two accounts
DTMF tones sending
Echo cancellation
Codec settings per account
Account password encryption
Call history 
Hold function
Quick dial panel
Optional Automatic pop-up window for incoming call
Call logs
Minimize on tray
Minimize on start up
Always on top
Adaptive Jitter Buffer
Support for multiple audio devices
Address book
Quickdial pad
Automatic user registration
Call transfer
Voice mail message information
Portable ZoIPer with portable devices (like USB, flashcards, etc.)
Multilanguage support</description>
            <homepage_url>http://www.zoiper.com/</homepage_url>
            <download_url>http://www.zoiper.com/free.php</download_url>
            <url_name>zoiper</url_name>
            <user_count>6</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id></analysis_id>
            <licenses>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238926</id>
          <stack_id>30353</stack_id>
          <project_id>16639</project_id>
          <created_at>2009-04-20T15:20:52Z</created_at>
          <project>
            <id>16639</id>
            <name>QuteCom</name>
            <created_at>2008-09-11T11:46:07Z</created_at>
            <updated_at>2009-12-22T07:51:57Z</updated_at>
            <description>QuteCom is the new name for the open source softphone previously known as WengoPhone.</description>
            <homepage_url>http://www.qutecom.org/</homepage_url>
            <download_url>http://www.qutecom.org/</download_url>
            <url_name>qutecom</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/18179/qutecom_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/18179/qutecom_small.png</small_logo_url>
            <user_count>5</user_count>
            <average_rating></average_rating>
            <rating_count>4</rating_count>
            <analysis_id>804124</analysis_id>
            <licenses>
              <license>
                <name>gpl2_or_later</name>
                <nice_name>GNU General Public License 2.0 or later</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>236974</id>
          <stack_id>30353</stack_id>
          <project_id>9287</project_id>
          <created_at>2009-04-15T11:04:22Z</created_at>
          <project>
            <id>9287</id>
            <name>SIP Router Project</name>
            <created_at>2007-10-23T08:57:44Z</created_at>
            <updated_at>2009-04-15T11:05:48Z</updated_at>
            <description>The SIP Router Project aims to create a common collaboration framework for all projects related to SIP Express Router (aka SER), referred further as x-SER projects. The projects that joined the initiative so far are: SIP Express Router (SER) - the initial project started in 2002 and Kamailio (OpenSER) - the project started as fork of SER in 2005.</description>
            <homepage_url>http://sip-router.org/</homepage_url>
            <download_url></download_url>
            <url_name></url_name>
            <user_count>1</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>510828</analysis_id>
            <licenses>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238709</id>
          <stack_id>30353</stack_id>
          <project_id>8366</project_id>
          <created_at>2009-04-19T21:12:36Z</created_at>
          <project>
            <id>8366</id>
            <name>Linphone</name>
            <created_at>2007-09-10T17:32:42Z</created_at>
            <updated_at>2009-12-24T04:18:12Z</updated_at>
            <description>Linphone is an internet phone or Voice Over IP phone (VoIP).</description>
            <homepage_url>http://www.linphone.org/index.php/eng</homepage_url>
            <download_url>http://www.linphone.org/index.php/eng/download</download_url>
            <url_name></url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/6065/linphone_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/6065/linphone_small.png</small_logo_url>
            <user_count>1</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>807304</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238710</id>
          <stack_id>30353</stack_id>
          <project_id>3980</project_id>
          <created_at>2009-04-19T21:14:15Z</created_at>
          <project>
            <id>3980</id>
            <name>WengoPhone</name>
            <created_at>2007-01-12T18:31:47Z</created_at>
            <updated_at>2009-05-16T02:56:14Z</updated_at>
            <description>OpenWengo is a community project focussed on communication over IP, including VoIP, instant messaging and video phonecalls. Its main product is the Wengophone - a standards-based softphone and multi-protocol IM client.

The successor of WengoPhone is QuteCom.</description>
            <homepage_url>http://www.wengophone.com/index.php</homepage_url>
            <download_url>http://www.wengophone.com/index.php/mp_download_wp_win</download_url>
            <url_name>openwengo</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/18155/wengophone-64_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/18155/wengophone-64_small.png</small_logo_url>
            <user_count>10</user_count>
            <average_rating>3.4</average_rating>
            <rating_count>4</rating_count>
            <analysis_id>560210</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238711</id>
          <stack_id>30353</stack_id>
          <project_id>797</project_id>
          <created_at>2009-04-19T21:14:34Z</created_at>
          <project>
            <id>797</id>
            <name>SIP Communicator</name>
            <created_at>2006-10-11T05:49:36Z</created_at>
            <updated_at>2009-12-24T09:50:07Z</updated_at>
            <description>SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular instant messaging and telephony protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, RSS, IRC and soon others like IAX.</description>
            <homepage_url>http://sip-communicator.org</homepage_url>
            <download_url>http://www.sip-communicator.org/index.php/Main/Download</download_url>
            <url_name>SIPCommunicator</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/4548/sip-communicator_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/4548/sip-communicator_small.png</small_logo_url>
            <user_count>12</user_count>
            <average_rating>4.4</average_rating>
            <rating_count>4</rating_count>
            <analysis_id>807645</analysis_id>
            <licenses>
              <license>
                <name>lgpl</name>
                <nice_name>GNU Lesser General Public License 2.1</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>229748</id>
          <stack_id>30353</stack_id>
          <project_id>317373</project_id>
          <created_at>2009-03-25T14:56:55Z</created_at>
          <project>
            <id>317373</id>
            <name>hangupd</name>
            <created_at>2009-02-26T09:26:46Z</created_at>
            <updated_at>2009-04-02T00:18:35Z</updated_at>
            <description>Simple companion to OpenSER-based family of SIP-servers, which is capable of hanging up calls.

It communicates with OpenSER/OpenSIPs/Kamailio with its FIFO interface, and can only close those calls, which are passed through OpenSER's dialog module.

It takes parameters of calls, which should be closed, from dedicated UnixODBC link, or receives them via native erlang messages.</description>
            <homepage_url>http://code.google.com/p/hangupd/</homepage_url>
            <download_url>http://code.google.com/p/hangupd/downloads/list</download_url>
            <url_name>hangupd</url_name>
            <user_count>3</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>503078</analysis_id>
            <licenses>
              <license>
                <name>gpl3_or_later</name>
                <nice_name>GNU General Public License 3 or later</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238093</id>
          <stack_id>30353</stack_id>
          <project_id>136838</project_id>
          <created_at>2009-04-17T20:05:12Z</created_at>
          <project>
            <id>136838</id>
            <name>cp30siprox</name>
            <created_at>2009-01-09T05:15:04Z</created_at>
            <updated_at>2009-01-09T05:15:05Z</updated_at>
            <description>The SIP Proxy server written in Java. Use www.cafesip.org Jiplet Container Architecture. It can be embedded with Python Script to maximize potentials</description>
            <homepage_url>http://code.google.com/p/cp30siprox</homepage_url>
            <download_url></download_url>
            <url_name>cp30siprox</url_name>
            <user_count>1</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id></analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238095</id>
          <stack_id>30353</stack_id>
          <project_id>65199</project_id>
          <created_at>2009-04-17T20:05:38Z</created_at>
          <project>
            <id>65199</id>
            <name>cvoip</name>
            <created_at>2008-12-18T15:08:27Z</created_at>
            <updated_at>2009-04-21T17:30:12Z</updated_at>
            <description>Projeto de 7&#186; periodo do Curso de Ci&#234;ncias da Computa&#231;&#227;o do Centro Universit&#225;rio do Par&#225;.</description>
            <homepage_url>http://code.google.com/p/cvoip</homepage_url>
            <download_url></download_url>
            <url_name>cvoip</url_name>
            <user_count>1</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>527693</analysis_id>
            <licenses>
              <license>
                <name>mozilla_public_1_1</name>
                <nice_name>Mozilla Public License 1.1</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238098</id>
          <stack_id>30353</stack_id>
          <project_id>326971</project_id>
          <created_at>2009-04-17T20:07:24Z</created_at>
          <project>
            <id>326971</id>
            <name>jiplet</name>
            <created_at>2009-04-17T20:07:18Z</created_at>
            <updated_at>2009-12-21T13:04:24Z</updated_at>
            <description>Jiplet is short for Java SIP Servlet. The Jiplet Container is an open-source container for server-side SIP applications. An application developer can create a SIP application written in Java using the Jiplet API and deploy the application in the container.</description>
            <homepage_url>http://www.cafesip.org/projects/jiplet/index.html</homepage_url>
            <download_url></download_url>
            <url_name>jiplet</url_name>
            <user_count>1</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id>802797</analysis_id>
            <licenses>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238157</id>
          <stack_id>30353</stack_id>
          <project_id>6765</project_id>
          <created_at>2009-04-17T20:24:29Z</created_at>
          <project>
            <id>6765</id>
            <name>Asterisk-Java</name>
            <created_at>2007-07-22T11:00:39Z</created_at>
            <updated_at>2009-12-16T17:27:51Z</updated_at>
            <description>Asterisk-Java, a free Java library for Asterisk PBX integration, consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server.

Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API.
The FastAGI implementation supports all commands currently available from Asterisk.
The Manager API implementation supports receiving events from the Asterisk server (e.g. call progess, registered peers, channel state) and sending actions to Asterisk (e.g. originate call, agent login/logoff, start/stop voice recording).</description>
            <homepage_url>http://asterisk-java.org</homepage_url>
            <download_url>http://asterisk-java.org</download_url>
            <url_name>freshmeat_asterisk-java</url_name>
            <user_count>10</user_count>
            <average_rating>4.8</average_rating>
            <rating_count>5</rating_count>
            <analysis_id>793835</analysis_id>
            <licenses>
              <license>
                <name>apache_2</name>
                <nice_name>Apache License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238158</id>
          <stack_id>30353</stack_id>
          <project_id>6766</project_id>
          <created_at>2009-04-17T20:25:08Z</created_at>
          <project>
            <id>6766</id>
            <name>Asterisk-IM</name>
            <created_at>2007-07-22T11:20:32Z</created_at>
            <updated_at>2009-02-13T15:55:57Z</updated_at>
            <description>Asterisk-IM is a plugin for the Openfire XMPP server that connects your phone with instant messaging.
Asterisk-IM manage your presence when you are on the phone. If you are using a client, that supports the corresponding XMPP extensions, you can also use Asterisk-IM to call your buddies and receive notifications on incoming calls.</description>
            <homepage_url>http://www.igniterealtime.org</homepage_url>
            <download_url>http://www.igniterealtime.org/projects/openfire/plugins.jsp</download_url>
            <url_name>AsteriskIM</url_name>
            <medium_logo_url>http://bits.ohloh.net/attachments/4536/asterisk_im_med.png</medium_logo_url>
            <small_logo_url>http://bits.ohloh.net/attachments/4536/asterisk_im_small.png</small_logo_url>
            <user_count>10</user_count>
            <average_rating></average_rating>
            <rating_count>4</rating_count>
            <analysis_id>206013</analysis_id>
            <licenses>
              <license>
                <name>gpl</name>
                <nice_name>GNU General Public License 2.0</nice_name>
              </license>
            </licenses>
          </project>
        </stack_entry>
        <stack_entry>
          <id>238159</id>
          <stack_id>30353</stack_id>
          <project_id>269647</project_id>
          <created_at>2009-04-17T20:25:32Z</created_at>
          <project>
            <id>269647</id>
            <name>Asterisk-GUI</name>
            <created_at>2009-01-21T00:36:57Z</created_at>
            <updated_at>2009-08-29T09:06:33Z</updated_at>
            <description>Asterisk GUI for Asterisk 1.2 Realtime using MySQLsupports IAX / SIP / FAX / Forward Calls / Outgoing Providers / Global Settings / Timeshedules for Voicemails and Ques</description>
            <homepage_url>http://sourceforge.net/projects/asterisk-gui</homepage_url>
            <download_url></download_url>
            <url_name>asterisk-gui</url_name>
            <user_count>3</user_count>
            <average_rating></average_rating>
            <rating_count>1</rating_count>
            <analysis_id></analysis_id>
            <licenses>
            </licenses>
          </project>
        </stack_entry>
      </stack_entries>
      <account_id>3283</account_id>
    </stack>
  </result>
</response>
