[Freeswitch-users] FS and Gtalk integration

[jesse zhao]

hi, help:

  I searched online pages and FS wiki , basically the doc about Gtalk
and FS is pretty lousy. I still couldn't get
my gtalk and FS work together.

   two major issues:

  1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established,
but no audio.
  2) how could I call SIP from gtalk ?  when I request chat to sip,
gchat in gmail requests an invitation.
     suppose the voice request is : conf+1000 at 172.18.115.73

 here are my config files:
 jingle_profiles/client.xml

  <profile type="client">
    <param name="name" value="gmail.com"/>
    <param name="login" value="XYZ at gmail.com/talk"/>
    <param name="password" value="XXXXX"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="md5"/>
    <!-- if the server where the jabber is hosted is not the same as
the one in the jid -->
    <param name="server" value="talk.google.com"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="jingle2sip"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>


 dialplan/default.xml

    <extension name="jingle2sip">
      <condition field="source" expression="mod_dingaling"/>
      <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$">
       <action application="bridge" data="sofia/default/$1%172.18.115.73"/>
      </condition>
    </extension>

    <extension name="sip2jingle">
          <condition field="source" expression="mod_sofia"/>
          <condition field="destination_number"
expression="^gtalk=([a-zA-z0-9.-]+)$">
            <action application="set" data="no_media=false"/>
            <action application="set"
data="effective_caller_id_number=myuser at gmail.com"/>
            <action application="bridge"
data="dingaling/gmail.com/$1 at gmail.com"/>
          </condition>

I tried suggestion from
http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456.html,
above two issues still not work.

 I heard id: intralanman is good at this, if you saw the post, please
share your advice.

-jesse