[Freeswitch-users] How to interface with google talk

[danish.samad at vocalseeds.com]

Hi,

I am trying to interface with google talk using the following setup

Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client

Freeswitch will register with Jabberserver using a valid jabber account on
the Jabber server. Using this account I should be able to login and add
any valid gmail id in its userlist and perform standard operations (using
the jingle protocol) such as chat, voice, etc. Eventually I intend to use
Freeswitch as a gateway between SIP and google talk clients, but I first
need to validate the above setup.

 To test the setup out, I created an account on jabber.anywise.com. I was
able to successfully log into the server using PSI and adding a valid
gmail accounts in my roster.  Now when I specify the account in
freeswitch.xml I am not able to register as I donot see the user coming
online in my google talk user list. I just see the following XML messages
being dumped on the console.

freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h
ttp://etherx.jabber.org/streams' xmlns='jabber:client'
to='jabber.anywise.com' version='1.0'>]

RECV[<?xml version='1.0'?><stream:stream
xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7'
xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>]

RECV[<stream:features><register
xmlns='http://jabber.org/features/iq-register'/><auth
xmlns='http://jabber.org/features/iq-auth'/></stream:features>]

For reference I am attaching related excerpts from the freeswitch.xml file

<load module="mod_dingaling"/>
....
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
      <settings>
        <param name="debug" value="1"/>
        <param name="codec-prefs" value="PCMU"/>
      </settings>
      <interface>
        <param name="name" value="jingle"/>
        <param name="login" value="user at jabber.anywise.com/talk"/>
        <param name="password" value="pwd"/>
        <param name="dialplan" value="XML"/>
        <param name="message" value="Jingle all the way"/>
        <param name="rtp-ip" value="vostroip"/>
        <param name="auto-login" value="true"/>
        <param name="use-rtp-timer" value="true"/>
	<param name="exten" value="7777"/>
        <param name="vad" value="both"/>
      </interface>
</configuration>
...

<extension name="7777">
 <condition field="destination_number" expression="^7777$">
   <action application="bridge" data="playback
/var/lib/asterisk/sounds/test.wav"/>
 </condition>
</extension>

Please suggest what I need to do to achieve the required setup. Any help
will be appreciated.

Regards,
Danish

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