git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3169 d0543943-73ff-0310-b7d9-9358b9ac24b2

Rework of Filter's handling of the socket with freeswitch, the previous implementation was a serious cheat.  We now properly handle the Content-Length tag and go out of line based mode and into length mode.

Also updated the console to show replies to commands.


git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3168 d0543943-73ff-0310-b7d9-9358b9ac24b2

move to new sofia tarball for windows build, this new tarball includes fix for build w/o glib on *nix, fix for segfault on cancel during early media, and proper implementation of handing subscribe and publish requests up to the application (removes our private hacks to enable this behavior).

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the <param> tags that was in the <user>
are now nested in thier own <params> tag.


Also when using dingaling in component mode you can add a <vcard> to the
user that will be deliverd when requested when using an id like 

user+stpeter@jabber.org assuming the comonent subdomain is jabber.org





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Initial checkin of the POE::Filter::FSSocket module and fsconsole.pl which is a curses, multi-window console for freeswitch mod_event_socket.  See http://search.cpan.org/~ptinsley/POE-Filter-FSSocket-0.04/lib/POE/Filter/FSSocket.pm for more info on the perl module.


git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3116 d0543943-73ff-0310-b7d9-9358b9ac24b2

This is some brand new stuff to gateway chat/presence/audio from one protocol to another
So far it only works between google/jingle and SIP

All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end.

With this setup registered X-Lite's can chat with each other and call each other 
as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls.

Chat May also be done between X-Lite and jabber 

You'll also need a jabber server configured for component login so you can interface.
We have only tested with jabberd2 so far.

Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example)
so the jabber records are pointed at your jabber server.

RELEVANT CONFIGS

<!-- Brian has no jingle support so send calls to him over to his iax url -->
<extension name="bkw">
  <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$">
    <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/>
  </condition>
</extension>

<!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below -->

<extension name="jingle2sip">
  <condition field="source" expression="mod_dingaling"/>
  <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$">
    <action application="bridge" data="sofia/$2/$1%$2"/>
  </condition>
</extension>

<extension name="sip2jingle">
  <condition field="source" expression="mod_sofia"/>
  <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$">
    <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/>
  </condition>
</extension>


<configuration name="sofia.conf" description="sofia Endpoint">
  <global_settings>
    <param name="log-level" value="0"/>
  </global_settings>

  <profiles>
    <profile name="fs.mydomain.com">
      <registrations/>
      <settings>
	<param name="debug" value="1"/>
	<param name="rfc2833-pt" value="101"/>
	<param name="sip-port" value="5060"/>
	<param name="dialplan" value="XML"/>
	<param name="dtmf-duration" value="100"/>
	<param name="codec-prefs" value="PCMU"/>
	<param name="codec-ms" value="20"/>
	<param name="accept-blind-reg" value="true"/>
	<param name="manage-presence" value="true"/>
	<!--<param name="full-id-in-dialplan" value="true"/>-->
	<!--<param name="auth-calls" value="true"/>-->
	<!--<param name="auth-all-packets" value="true"/>-->
	<param name="use-rtp-timer" value="true"/>
	<param name="rtp-timer-name" value="soft"/>
	<param name="rtp-ip" value="100.200.100.200"/>
	<param name="sip-ip" value="fs.mydomain.com"/>
      </settings>
    </profile>
  </profiles>

</configuration>


<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
  <settings>
    <param name="debug" value="0"/>
    <param name="codec-prefs" value="PCMU"/>
  </settings>

  <profile type="component">
    <param name="name" value="fs.mydomain.com"/>
    <param name="password" value="secret"/>
    <param name="dialplan" value="XML"/>
    <param name="rtp-ip" value="208.64.200.42"/>
    <param name="server" value="jabber.freeswitch.org:5347"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <param name="exten" value="_auto_"/>
    <!--<param name="vad" value="both"/>-->
  </profile>

</configuration>



git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3114 d0543943-73ff-0310-b7d9-9358b9ac24b2

Fix for broken trunk commit, setting contact_str size to the size of sql variable it's stored in [contact VARCHAR(1024)] + 1 for the null.


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