here my setups that work as an asterisk-to-gtk gateway. it works perfectly when connecting to/from original gtalk on windows but not with linux clients as jabbin or psi or tapioca. in those cases the codec negociation fails somehow...

here my dingaling.conf:

   <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
      <settings>
        <param name="debug" value="1"/>
        <param name="codec-prefs" value="PCMU@8000"/>
      </settings>

     <profile type="client">
        <param name="name" value="gtalk"/>
        <param name="login" value="myuser@gmail.com/talk"/>
        <param name="password" value="mypasswd"/>
        <param name="dialplan" value="XML"/>
        <param name="message" value="Jingle-to-SIP Gateway via Freeswitch"/>
        <param name="rtp-ip" value="85.126.xxx.xxx"/>
        <param name="auto-login" value="true"/>
        <param name="auto-reply" value="Press *Call* to call me via FreeSWITCH and visit me at http://pekka.kurki.at"/>
        <!-- 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:5222"/>
        <!-- Enable TLS or not -->
        <param name="tls" value="true"/>
        <!-- disable to trade async for more calls -->
        <param name="use-rtp-timer" value="true"/>
        <!-- or -->
        <!-- <param name="rtp-ip" value="my_lan_ip"/> -->
        <param name="ext-rtp-ip" value="85.126.xxx.xxx"/>
        <!-- default extension (if one cannot be determined) -->
        <param name="exten" value="1004j"/>
        <!-- VAD choose one -->
        <!-- <param name="vad" value="in"/> -->
        <!-- <param name="vad" value="out"/> -->
        <param name="vad" value="both"/>
      </profile>
   </configuration>

and here the relevant parts of default.xml:

<context name="default">

      <extension name="gtalk">
          <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@gmail.com"/>
            <action application="bridge" data="dingaling/gtalk/$1@gmail.com"/>
          </condition>
      </extension>

      <extension name="1004j">
          <condition field="source" expression="mod_dingaling"/>
          <condition field="destination_number" expression="^1004j">
             <action application="set" data="no_media=false"/>
             <action application="bridge" data="sofia/sip.intellectics.com/mysipuser@sip.intellectics.com"/>
          </condition>
      </extension>

</context>

- sip calls with target address "gtalk=usrname" are forwarded to  gtalk
- all calls from gtalk are forwarded to my sip phone on asterisk server

this works ok for me but i am still looking for solution for linux clients...

br,

pekka


Shiladitya Sircar wrote:
Hello,
I was wondering if any of you can point me towards some documentation in configuring FreeSwitch with Google Talk, using dingaling.conf ? A sample dingaling.conf and a dail plan would be very helpful.

thanks
-ssircar

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-- 
(Mr.) Pekka K. Kurki
Intellectics GesmbH
http://www.intellectics.com
Helenenstrasse 44, A-2500 Baden, Austria
Tel. +43 2256 815811  Mobile +43 699 12532539 
Fax  +43 2256 815819
(Tel. US: 360-2267729 UK: 0870 4787064)
sip:pekka.kurki@sip.intellectics.com
skype: pkkurki, pkkurki-sip (sip gateway)
gtalk: pekkis50@gtalk.com (sip gateway - experimental)
gizmo: pekkis50
h323:432256815811@sip.intellectics.com
email: Pekka.Kurki@intellectics.com
www: http://pekka.kurki.at